2 * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
3 * <2006> Lutz Mueller <lutz at topfrose dot de>
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
21 * Unless otherwise indicated, Source Code is licensed under MIT license.
22 * See further explanation attached in License Statement (distributed in the file
25 * Permission is hereby granted, free of charge, to any person obtaining a copy of
26 * this software and associated documentation files (the "Software"), to deal in
27 * the Software without restriction, including without limitation the rights to
28 * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
29 * of the Software, and to permit persons to whom the Software is furnished to do
30 * so, subject to the following conditions:
32 * The above copyright notice and this permission notice shall be included in all
33 * copies or substantial portions of the Software.
35 * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
36 * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
37 * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
38 * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
39 * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
40 * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
44 * SECTION:element-rtspsrc
46 * Makes a connection to an RTSP server and read the data.
47 * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
48 * RealMedia/Quicktime/Microsoft extensions.
50 * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
51 * default rtspsrc will negotiate a connection in the following order:
52 * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
53 * protocols can be controlled with the #GstRTSPSrc:protocols property.
55 * rtspsrc currently understands SDP as the format of the session description.
56 * For each stream listed in the SDP a new rtp_stream\%d pad will be created
57 * with caps derived from the SDP media description. This is a caps of mime type
58 * "application/x-rtp" that can be connected to any available RTP depayloader
61 * rtspsrc will internally instantiate an RTP session manager element
62 * that will handle the RTCP messages to and from the server, jitter removal,
63 * packet reordering along with providing a clock for the pipeline.
64 * This feature is implemented using the gstrtpbin element.
66 * rtspsrc acts like a live source and will therefore only generate data in the
70 * <title>Example launch line</title>
72 * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
73 * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
84 #endif /* HAVE_UNISTD_H */
90 #include <gst/net/gstnet.h>
91 #include <gst/sdp/gstsdpmessage.h>
92 #include <gst/sdp/gstmikey.h>
93 #include <gst/rtp/rtp.h>
95 #include "gst/gst-i18n-plugin.h"
97 #include "gstrtspsrc.h"
99 GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
100 #define GST_CAT_DEFAULT (rtspsrc_debug)
102 static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
105 GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
107 /* templates used internally */
108 static GstStaticPadTemplate anysrctemplate =
109 GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
112 GST_STATIC_CAPS_ANY);
114 static GstStaticPadTemplate anysinktemplate =
115 GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
118 GST_STATIC_CAPS_ANY);
122 SIGNAL_HANDLE_REQUEST,
124 SIGNAL_SELECT_STREAM,
126 SIGNAL_REQUEST_RTCP_KEY,
130 enum _GstRtspSrcRtcpSyncMode
137 enum _GstRtspSrcBufferMode
146 #define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
148 gst_rtsp_src_buffer_mode_get_type (void)
150 static GType buffer_mode_type = 0;
151 static const GEnumValue buffer_modes[] = {
152 {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
153 {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
154 {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
155 {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
156 {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
160 if (!buffer_mode_type) {
162 g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
164 return buffer_mode_type;
167 #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
238 #define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
240 gst_rtsp_nat_method_get_type (void)
242 static GType rtsp_nat_method_type = 0;
243 static const GEnumValue rtsp_nat_method[] = {
244 {GST_RTSP_NAT_NONE, "None", "none"},
245 {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
249 if (!rtsp_nat_method_type) {
250 rtsp_nat_method_type =
251 g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
253 return rtsp_nat_method_type;
256 static void gst_rtspsrc_finalize (GObject * object);
258 static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
259 const GValue * value, GParamSpec * pspec);
260 static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
261 GValue * value, GParamSpec * pspec);
263 static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
265 static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
266 gpointer iface_data);
268 static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
271 static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
272 static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
274 static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
276 static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
277 GstStateChange transition);
278 static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
279 static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
281 static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
282 GstRTSPMessage * response);
284 static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
286 static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
287 GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
289 static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
290 static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
292 static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
293 static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
294 gboolean only_close);
296 static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
297 const gchar * uri, GError ** error);
298 static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
300 static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
301 static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
302 static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
303 GstRTSPStream * stream, GstEvent * event);
304 static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
305 static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
313 /* commands we send to out loop to notify it of events */
314 #define CMD_OPEN (1 << 0)
315 #define CMD_PLAY (1 << 1)
316 #define CMD_PAUSE (1 << 2)
317 #define CMD_CLOSE (1 << 3)
318 #define CMD_WAIT (1 << 4)
319 #define CMD_RECONNECT (1 << 5)
320 #define CMD_LOOP (1 << 6)
322 /* mask for all commands */
323 #define CMD_ALL ((CMD_LOOP << 1) - 1)
325 #define GST_ELEMENT_PROGRESS(el, type, code, text) \
327 gchar *__txt = _gst_element_error_printf text; \
328 gst_element_post_message (GST_ELEMENT_CAST (el), \
329 gst_message_new_progress (GST_OBJECT_CAST (el), \
330 GST_PROGRESS_TYPE_ ##type, code, __txt)); \
334 static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
336 #define gst_rtspsrc_parent_class parent_class
337 G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
338 G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
340 #ifndef GST_DISABLE_GST_DEBUG
341 static inline const char *
342 cmd_to_string (guint cmd)
366 default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
368 GST_DEBUG_OBJECT (src, "default handler");
373 select_stream_accum (GSignalInvocationHint * ihint,
374 GValue * return_accu, const GValue * handler_return, gpointer data)
378 myboolean = g_value_get_boolean (handler_return);
379 GST_DEBUG ("accum %d", myboolean);
380 g_value_set_boolean (return_accu, myboolean);
382 /* stop emission if FALSE */
387 gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
389 GObjectClass *gobject_class;
390 GstElementClass *gstelement_class;
391 GstBinClass *gstbin_class;
393 gobject_class = (GObjectClass *) klass;
394 gstelement_class = (GstElementClass *) klass;
395 gstbin_class = (GstBinClass *) klass;
397 GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
399 gobject_class->set_property = gst_rtspsrc_set_property;
400 gobject_class->get_property = gst_rtspsrc_get_property;
402 gobject_class->finalize = gst_rtspsrc_finalize;
404 g_object_class_install_property (gobject_class, PROP_LOCATION,
405 g_param_spec_string ("location", "RTSP Location",
406 "Location of the RTSP url to read",
407 DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
409 g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
410 g_param_spec_flags ("protocols", "Protocols",
411 "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
412 DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
414 g_object_class_install_property (gobject_class, PROP_DEBUG,
415 g_param_spec_boolean ("debug", "Debug",
416 "Dump request and response messages to stdout",
417 DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
419 g_object_class_install_property (gobject_class, PROP_RETRY,
420 g_param_spec_uint ("retry", "Retry",
421 "Max number of retries when allocating RTP ports.",
422 0, G_MAXUINT16, DEFAULT_RETRY,
423 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
425 g_object_class_install_property (gobject_class, PROP_TIMEOUT,
426 g_param_spec_uint64 ("timeout", "Timeout",
427 "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
428 0, G_MAXUINT64, DEFAULT_TIMEOUT,
429 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
431 g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
432 g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
433 "Fail after timeout microseconds on TCP connections (0 = disabled)",
434 0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
435 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
437 g_object_class_install_property (gobject_class, PROP_LATENCY,
438 g_param_spec_uint ("latency", "Buffer latency in ms",
439 "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
440 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
442 g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
443 g_param_spec_boolean ("drop-on-latency",
444 "Drop buffers when maximum latency is reached",
445 "Tells the jitterbuffer to never exceed the given latency in size",
446 DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
448 g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
449 g_param_spec_uint64 ("connection-speed", "Connection Speed",
450 "Network connection speed in kbps (0 = unknown)",
451 0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
452 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
454 g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
455 g_param_spec_enum ("nat-method", "NAT Method",
456 "Method to use for traversing firewalls and NAT",
457 GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
458 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
461 * GstRTSPSrc:do-rtcp:
463 * Enable RTCP support. Some old server don't like RTCP and then this property
464 * needs to be set to FALSE.
466 g_object_class_install_property (gobject_class, PROP_DO_RTCP,
467 g_param_spec_boolean ("do-rtcp", "Do RTCP",
468 "Send RTCP packets, disable for old incompatible server.",
469 DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
472 * GstRTSPSrc:do-rtsp-keep-alive:
474 * Enable RTSP keep alive support. Some old server don't like RTSP
475 * keep alive and then this property needs to be set to FALSE.
477 g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
478 g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
479 "Send RTSP keep alive packets, disable for old incompatible server.",
480 DEFAULT_DO_RTSP_KEEP_ALIVE,
481 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
486 * Set the proxy parameters. This has to be a string of the format
487 * [http://][user:passwd@]host[:port].
489 g_object_class_install_property (gobject_class, PROP_PROXY,
490 g_param_spec_string ("proxy", "Proxy",
491 "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
492 DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
494 * GstRTSPSrc:proxy-id:
496 * Sets the proxy URI user id for authentication. If the URI set via the
497 * "proxy" property contains a user-id already, that will take precedence.
501 g_object_class_install_property (gobject_class, PROP_PROXY_ID,
502 g_param_spec_string ("proxy-id", "proxy-id",
503 "HTTP proxy URI user id for authentication", "",
504 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
506 * GstRTSPSrc:proxy-pw:
508 * Sets the proxy URI password for authentication. If the URI set via the
509 * "proxy" property contains a password already, that will take precedence.
513 g_object_class_install_property (gobject_class, PROP_PROXY_PW,
514 g_param_spec_string ("proxy-pw", "proxy-pw",
515 "HTTP proxy URI user password for authentication", "",
516 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
519 * GstRTSPSrc:rtp-blocksize:
521 * RTP package size to suggest to server.
523 g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
524 g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
525 "RTP package size to suggest to server (0 = disabled)",
526 0, 65536, DEFAULT_RTP_BLOCKSIZE,
527 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
529 g_object_class_install_property (gobject_class,
531 g_param_spec_string ("user-id", "user-id",
532 "RTSP location URI user id for authentication", DEFAULT_USER_ID,
533 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
534 g_object_class_install_property (gobject_class, PROP_USER_PW,
535 g_param_spec_string ("user-pw", "user-pw",
536 "RTSP location URI user password for authentication", DEFAULT_USER_PW,
537 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
540 * GstRTSPSrc:buffer-mode:
542 * Control the buffering and timestamping mode used by the jitterbuffer.
544 g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
545 g_param_spec_enum ("buffer-mode", "Buffer Mode",
546 "Control the buffering algorithm in use",
547 GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
548 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
551 * GstRTSPSrc:port-range:
553 * Configure the client port numbers that can be used to recieve RTP and
556 g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
557 g_param_spec_string ("port-range", "Port range",
558 "Client port range that can be used to receive RTP and RTCP data, "
559 "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
560 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
563 * GstRTSPSrc:udp-buffer-size:
565 * Size of the kernel UDP receive buffer in bytes.
567 g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
568 g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
569 "Size of the kernel UDP receive buffer in bytes, 0=default",
570 0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
571 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
574 * GstRTSPSrc:short-header:
576 * Only send the basic RTSP headers for broken encoders.
578 g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
579 g_param_spec_boolean ("short-header", "Short Header",
580 "Only send the basic RTSP headers for broken encoders",
581 DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
583 g_object_class_install_property (gobject_class, PROP_PROBATION,
584 g_param_spec_uint ("probation", "Number of probations",
585 "Consecutive packet sequence numbers to accept the source",
586 0, G_MAXUINT, DEFAULT_PROBATION,
587 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
589 g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
590 g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
591 "Reconnect to the server if RTSP connection is closed when doing UDP",
592 DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
594 g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
595 g_param_spec_string ("multicast-iface", "Multicast Interface",
596 "The network interface on which to join the multicast group",
597 DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
599 g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
600 g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
601 "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
602 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
604 g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
605 g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
606 "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
607 DEFAULT_USE_PIPELINE_CLOCK,
608 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
610 g_object_class_install_property (gobject_class, PROP_SDES,
611 g_param_spec_boxed ("sdes", "SDES",
612 "The SDES items of this session",
613 GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
616 * GstRTSPSrc::tls-validation-flags:
618 * TLS certificate validation flags used to validate server
623 g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
624 g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
625 "TLS certificate validation flags used to validate the server certificate",
626 G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
627 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
630 * GstRTSPSrc::tls-database:
632 * TLS database with anchor certificate authorities used to validate
633 * the server certificate.
637 g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
638 g_param_spec_object ("tls-database", "TLS database",
639 "TLS database with anchor certificate authorities used to validate the server certificate",
640 G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
643 * GstRTSPSrc::do-retransmission:
645 * Attempt to ask the server to retransmit lost packets according to RFC4588.
647 * Note: currently only works with SSRC-multiplexed retransmission streams
651 g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
652 g_param_spec_boolean ("do-retransmission", "Retransmission",
653 "Ask the server to retransmit lost packets",
654 DEFAULT_DO_RETRANSMISSION,
655 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
658 * GstRTSPSrc::handle-request:
659 * @rtspsrc: a #GstRTSPSrc
660 * @request: a #GstRTSPMessage
661 * @response: a #GstRTSPMessage
663 * Handle a server request in @request and prepare @response.
665 * This signal is called from the streaming thread, you should therefore not
666 * do any state changes on @rtspsrc because this might deadlock. If you want
667 * to modify the state as a result of this signal, post a
668 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
673 gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
674 g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
675 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
676 G_TYPE_POINTER, G_TYPE_POINTER);
679 * GstRTSPSrc::on-sdp:
680 * @rtspsrc: a #GstRTSPSrc
681 * @sdp: a #GstSDPMessage
683 * Emited when the client has retrieved the SDP and before it configures the
684 * streams in the SDP. @sdp can be inspected and modified.
686 * This signal is called from the streaming thread, you should therefore not
687 * do any state changes on @rtspsrc because this might deadlock. If you want
688 * to modify the state as a result of this signal, post a
689 * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
694 gst_rtspsrc_signals[SIGNAL_ON_SDP] =
695 g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
696 0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
697 GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
700 * GstRTSPSrc::select-stream:
701 * @rtspsrc: a #GstRTSPSrc
702 * @num: the stream number
703 * @caps: the stream caps
705 * Emited before the client decides to configure the stream @num with
708 * Returns: %TRUE when the stream should be selected, %FALSE when the stream
713 gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
714 g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
715 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
716 (GCallback) default_select_stream, select_stream_accum, NULL,
717 g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
720 * GstRTSPSrc::new-manager:
721 * @rtspsrc: a #GstRTSPSrc
722 * @manager: a #GstElement
724 * Emited after a new manager (like rtpbin) was created and the default
725 * properties were configured.
729 gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
730 g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
731 G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
732 g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
735 * GstRTSPSrc::request-rtcp-key:
736 * @rtspsrc: a #GstRTSPSrc
737 * @num: the stream number
739 * Signal emited to get the crypto parameters relevant to the RTCP
740 * stream. User should provide the key and the RTCP encryption ciphers
741 * and authentication, and return them wrapped in a GstCaps.
745 gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY] =
746 g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
747 G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
749 gstelement_class->send_event = gst_rtspsrc_send_event;
750 gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
751 gstelement_class->change_state = gst_rtspsrc_change_state;
753 gst_element_class_add_pad_template (gstelement_class,
754 gst_static_pad_template_get (&rtptemplate));
756 gst_element_class_set_static_metadata (gstelement_class,
757 "RTSP packet receiver", "Source/Network",
758 "Receive data over the network via RTSP (RFC 2326)",
759 "Wim Taymans <wim@fluendo.com>, "
760 "Thijs Vermeir <thijs.vermeir@barco.com>, "
761 "Lutz Mueller <lutz@topfrose.de>");
763 gstbin_class->handle_message = gst_rtspsrc_handle_message;
765 gst_rtsp_ext_list_init ();
769 gst_rtspsrc_init (GstRTSPSrc * src)
771 src->conninfo.location = g_strdup (DEFAULT_LOCATION);
772 src->protocols = DEFAULT_PROTOCOLS;
773 src->debug = DEFAULT_DEBUG;
774 src->retry = DEFAULT_RETRY;
775 src->udp_timeout = DEFAULT_TIMEOUT;
776 gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
777 src->latency = DEFAULT_LATENCY_MS;
778 src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
779 src->connection_speed = DEFAULT_CONNECTION_SPEED;
780 src->nat_method = DEFAULT_NAT_METHOD;
781 src->do_rtcp = DEFAULT_DO_RTCP;
782 src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
783 gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
784 src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
785 src->user_id = g_strdup (DEFAULT_USER_ID);
786 src->user_pw = g_strdup (DEFAULT_USER_PW);
787 src->buffer_mode = DEFAULT_BUFFER_MODE;
788 src->client_port_range.min = 0;
789 src->client_port_range.max = 0;
790 src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
791 src->short_header = DEFAULT_SHORT_HEADER;
792 src->probation = DEFAULT_PROBATION;
793 src->udp_reconnect = DEFAULT_UDP_RECONNECT;
794 src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
795 src->ntp_sync = DEFAULT_NTP_SYNC;
796 src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
798 src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
799 src->tls_database = DEFAULT_TLS_DATABASE;
800 src->do_retransmission = DEFAULT_DO_RETRANSMISSION;
802 /* get a list of all extensions */
803 src->extensions = gst_rtsp_ext_list_get ();
805 /* connect to send signal */
806 gst_rtsp_ext_list_connect (src->extensions, "send",
807 (GCallback) gst_rtspsrc_send_cb, src);
809 /* protects the streaming thread in interleaved mode or the polling
810 * thread in UDP mode. */
811 g_rec_mutex_init (&src->stream_rec_lock);
813 /* protects our state changes from multiple invocations */
814 g_rec_mutex_init (&src->state_rec_lock);
816 src->state = GST_RTSP_STATE_INVALID;
818 GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
822 gst_rtspsrc_finalize (GObject * object)
826 rtspsrc = GST_RTSPSRC (object);
828 gst_rtsp_ext_list_free (rtspsrc->extensions);
829 g_free (rtspsrc->conninfo.location);
830 gst_rtsp_url_free (rtspsrc->conninfo.url);
831 g_free (rtspsrc->conninfo.url_str);
832 g_free (rtspsrc->user_id);
833 g_free (rtspsrc->user_pw);
834 g_free (rtspsrc->multi_iface);
837 gst_sdp_message_free (rtspsrc->sdp);
840 if (rtspsrc->provided_clock)
841 gst_object_unref (rtspsrc->provided_clock);
844 gst_structure_free (rtspsrc->sdes);
846 if (rtspsrc->tls_database)
847 g_object_unref (rtspsrc->tls_database);
850 g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
851 g_rec_mutex_clear (&rtspsrc->state_rec_lock);
853 G_OBJECT_CLASS (parent_class)->finalize (object);
857 gst_rtspsrc_provide_clock (GstElement * element)
859 GstRTSPSrc *src = GST_RTSPSRC (element);
862 if ((clock = src->provided_clock) != NULL)
863 gst_object_ref (clock);
868 /* a proxy string of the format [user:passwd@]host[:port] */
870 gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
874 g_free (rtsp->proxy_user);
875 rtsp->proxy_user = NULL;
876 g_free (rtsp->proxy_passwd);
877 rtsp->proxy_passwd = NULL;
878 g_free (rtsp->proxy_host);
879 rtsp->proxy_host = NULL;
880 rtsp->proxy_port = 0;
887 /* we allow http:// in front but ignore it */
888 if (g_str_has_prefix (p, "http://"))
891 at = strchr (p, '@');
893 /* look for user:passwd */
894 col = strchr (proxy, ':');
895 if (col == NULL || col > at)
898 rtsp->proxy_user = g_strndup (p, col - p);
900 rtsp->proxy_passwd = g_strndup (col, at - col);
905 if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
906 rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
907 if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
908 rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
909 if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
910 GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
911 GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
914 col = strchr (p, ':');
917 /* everything before the colon is the hostname */
918 rtsp->proxy_host = g_strndup (p, col - p);
920 rtsp->proxy_port = strtoul (p, (char **) &p, 10);
922 rtsp->proxy_host = g_strdup (p);
923 rtsp->proxy_port = 8080;
929 gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
931 rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
932 rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
935 rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
937 rtspsrc->ptcp_timeout = NULL;
941 gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
946 rtspsrc = GST_RTSPSRC (object);
950 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
951 g_value_get_string (value), NULL);
954 rtspsrc->protocols = g_value_get_flags (value);
957 rtspsrc->debug = g_value_get_boolean (value);
960 rtspsrc->retry = g_value_get_uint (value);
963 rtspsrc->udp_timeout = g_value_get_uint64 (value);
965 case PROP_TCP_TIMEOUT:
966 gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
969 rtspsrc->latency = g_value_get_uint (value);
971 case PROP_DROP_ON_LATENCY:
972 rtspsrc->drop_on_latency = g_value_get_boolean (value);
974 case PROP_CONNECTION_SPEED:
975 rtspsrc->connection_speed = g_value_get_uint64 (value);
977 case PROP_NAT_METHOD:
978 rtspsrc->nat_method = g_value_get_enum (value);
981 rtspsrc->do_rtcp = g_value_get_boolean (value);
983 case PROP_DO_RTSP_KEEP_ALIVE:
984 rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
987 gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
990 if (rtspsrc->prop_proxy_id)
991 g_free (rtspsrc->prop_proxy_id);
992 rtspsrc->prop_proxy_id = g_value_dup_string (value);
995 if (rtspsrc->prop_proxy_pw)
996 g_free (rtspsrc->prop_proxy_pw);
997 rtspsrc->prop_proxy_pw = g_value_dup_string (value);
999 case PROP_RTP_BLOCKSIZE:
1000 rtspsrc->rtp_blocksize = g_value_get_uint (value);
1003 if (rtspsrc->user_id)
1004 g_free (rtspsrc->user_id);
1005 rtspsrc->user_id = g_value_dup_string (value);
1008 if (rtspsrc->user_pw)
1009 g_free (rtspsrc->user_pw);
1010 rtspsrc->user_pw = g_value_dup_string (value);
1012 case PROP_BUFFER_MODE:
1013 rtspsrc->buffer_mode = g_value_get_enum (value);
1015 case PROP_PORT_RANGE:
1019 str = g_value_get_string (value);
1021 sscanf (str, "%u-%u",
1022 &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
1024 rtspsrc->client_port_range.min = 0;
1025 rtspsrc->client_port_range.max = 0;
1029 case PROP_UDP_BUFFER_SIZE:
1030 rtspsrc->udp_buffer_size = g_value_get_int (value);
1032 case PROP_SHORT_HEADER:
1033 rtspsrc->short_header = g_value_get_boolean (value);
1035 case PROP_PROBATION:
1036 rtspsrc->probation = g_value_get_uint (value);
1038 case PROP_UDP_RECONNECT:
1039 rtspsrc->udp_reconnect = g_value_get_boolean (value);
1041 case PROP_MULTICAST_IFACE:
1042 g_free (rtspsrc->multi_iface);
1044 if (g_value_get_string (value) == NULL)
1045 rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
1047 rtspsrc->multi_iface = g_value_dup_string (value);
1050 rtspsrc->ntp_sync = g_value_get_boolean (value);
1052 case PROP_USE_PIPELINE_CLOCK:
1053 rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
1056 rtspsrc->sdes = g_value_dup_boxed (value);
1058 case PROP_TLS_VALIDATION_FLAGS:
1059 rtspsrc->tls_validation_flags = g_value_get_flags (value);
1061 case PROP_TLS_DATABASE:
1062 g_clear_object (&rtspsrc->tls_database);
1063 rtspsrc->tls_database = g_value_dup_object (value);
1065 case PROP_DO_RETRANSMISSION:
1066 rtspsrc->do_retransmission = g_value_get_boolean (value);
1069 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1075 gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
1078 GstRTSPSrc *rtspsrc;
1080 rtspsrc = GST_RTSPSRC (object);
1084 g_value_set_string (value, rtspsrc->conninfo.location);
1086 case PROP_PROTOCOLS:
1087 g_value_set_flags (value, rtspsrc->protocols);
1090 g_value_set_boolean (value, rtspsrc->debug);
1093 g_value_set_uint (value, rtspsrc->retry);
1096 g_value_set_uint64 (value, rtspsrc->udp_timeout);
1098 case PROP_TCP_TIMEOUT:
1102 timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
1103 rtspsrc->tcp_timeout.tv_usec;
1104 g_value_set_uint64 (value, timeout);
1108 g_value_set_uint (value, rtspsrc->latency);
1110 case PROP_DROP_ON_LATENCY:
1111 g_value_set_boolean (value, rtspsrc->drop_on_latency);
1113 case PROP_CONNECTION_SPEED:
1114 g_value_set_uint64 (value, rtspsrc->connection_speed);
1116 case PROP_NAT_METHOD:
1117 g_value_set_enum (value, rtspsrc->nat_method);
1120 g_value_set_boolean (value, rtspsrc->do_rtcp);
1122 case PROP_DO_RTSP_KEEP_ALIVE:
1123 g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
1129 if (rtspsrc->proxy_host) {
1131 g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
1135 g_value_take_string (value, str);
1139 g_value_set_string (value, rtspsrc->prop_proxy_id);
1142 g_value_set_string (value, rtspsrc->prop_proxy_pw);
1144 case PROP_RTP_BLOCKSIZE:
1145 g_value_set_uint (value, rtspsrc->rtp_blocksize);
1148 g_value_set_string (value, rtspsrc->user_id);
1151 g_value_set_string (value, rtspsrc->user_pw);
1153 case PROP_BUFFER_MODE:
1154 g_value_set_enum (value, rtspsrc->buffer_mode);
1156 case PROP_PORT_RANGE:
1160 if (rtspsrc->client_port_range.min != 0) {
1161 str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
1162 rtspsrc->client_port_range.max);
1166 g_value_take_string (value, str);
1169 case PROP_UDP_BUFFER_SIZE:
1170 g_value_set_int (value, rtspsrc->udp_buffer_size);
1172 case PROP_SHORT_HEADER:
1173 g_value_set_boolean (value, rtspsrc->short_header);
1175 case PROP_PROBATION:
1176 g_value_set_uint (value, rtspsrc->probation);
1178 case PROP_UDP_RECONNECT:
1179 g_value_set_boolean (value, rtspsrc->udp_reconnect);
1181 case PROP_MULTICAST_IFACE:
1182 g_value_set_string (value, rtspsrc->multi_iface);
1185 g_value_set_boolean (value, rtspsrc->ntp_sync);
1187 case PROP_USE_PIPELINE_CLOCK:
1188 g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
1191 g_value_set_boxed (value, rtspsrc->sdes);
1193 case PROP_TLS_VALIDATION_FLAGS:
1194 g_value_set_flags (value, rtspsrc->tls_validation_flags);
1196 case PROP_TLS_DATABASE:
1197 g_value_set_object (value, rtspsrc->tls_database);
1199 case PROP_DO_RETRANSMISSION:
1200 g_value_set_boolean (value, rtspsrc->do_retransmission);
1203 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
1209 find_stream_by_id (GstRTSPStream * stream, gint * id)
1211 if (stream->id == *id)
1218 find_stream_by_channel (GstRTSPStream * stream, gint * channel)
1220 if (stream->channel[0] == *channel || stream->channel[1] == *channel)
1227 find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
1229 GstElement *src = (GstElement *) a;
1231 if (stream->udpsrc[0] == src)
1233 if (stream->udpsrc[1] == src)
1240 find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
1242 if (stream->conninfo.location) {
1243 /* check qualified setup_url */
1244 if (!strcmp (stream->conninfo.location, (gchar *) a))
1247 if (stream->control_url) {
1248 /* check original control_url */
1249 if (!strcmp (stream->control_url, (gchar *) a))
1252 /* check if qualified setup_url ends with string */
1253 if (g_str_has_suffix (stream->control_url, (gchar *) a))
1260 static GstRTSPStream *
1261 find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
1265 /* find and get stream */
1266 if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
1267 return (GstRTSPStream *) lstream->data;
1272 static const GstSDPBandwidth *
1273 gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1274 const GstSDPMedia * media, const gchar * type)
1278 /* first look in the media specific section */
1279 len = gst_sdp_media_bandwidths_len (media);
1280 for (i = 0; i < len; i++) {
1281 const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
1283 if (strcmp (bw->bwtype, type) == 0)
1286 /* then look in the message specific section */
1287 len = gst_sdp_message_bandwidths_len (sdp);
1288 for (i = 0; i < len; i++) {
1289 const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
1291 if (strcmp (bw->bwtype, type) == 0)
1298 gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
1299 const GstSDPMedia * media, GstRTSPStream * stream)
1301 const GstSDPBandwidth *bw;
1303 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
1304 stream->as_bandwidth = bw->bandwidth;
1306 stream->as_bandwidth = -1;
1308 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
1309 stream->rr_bandwidth = bw->bandwidth;
1311 stream->rr_bandwidth = -1;
1313 if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
1314 stream->rs_bandwidth = bw->bandwidth;
1316 stream->rs_bandwidth = -1;
1320 gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
1321 const GstSDPConnection * conn)
1323 if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
1326 if (conn->addrtype == NULL)
1329 /* check for IPV6 */
1330 if (strcmp (conn->addrtype, "IP4") == 0)
1331 stream->is_ipv6 = FALSE;
1332 else if (strcmp (conn->addrtype, "IP6") == 0)
1333 stream->is_ipv6 = TRUE;
1338 g_free (stream->destination);
1339 stream->destination = g_strdup (conn->address);
1341 /* check for multicast */
1342 stream->is_multicast =
1343 gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
1345 stream->ttl = conn->ttl;
1348 /* Go over the connections for a stream.
1349 * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
1351 * - If we are dealing with a localhost address, we disable multicast
1354 gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
1355 const GstSDPMedia * media, GstRTSPStream * stream)
1357 const GstSDPConnection *conn;
1360 /* first look in the media specific section */
1361 len = gst_sdp_media_connections_len (media);
1362 for (i = 0; i < len; i++) {
1363 conn = gst_sdp_media_get_connection (media, i);
1365 gst_rtspsrc_do_stream_connection (src, stream, conn);
1367 /* then look in the message specific section */
1368 if ((conn = gst_sdp_message_get_connection (sdp))) {
1369 gst_rtspsrc_do_stream_connection (src, stream, conn);
1373 /* m=<media> <UDP port> RTP/AVP <payload>
1376 gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
1377 const GstSDPMedia * media, GstRTSPStream * stream)
1383 proto = gst_sdp_media_get_proto (media);
1387 if (g_str_equal (proto, "RTP/AVP"))
1388 stream->profile = GST_RTSP_PROFILE_AVP;
1389 else if (g_str_equal (proto, "RTP/SAVP"))
1390 stream->profile = GST_RTSP_PROFILE_SAVP;
1391 else if (g_str_equal (proto, "RTP/AVPF"))
1392 stream->profile = GST_RTSP_PROFILE_AVPF;
1393 else if (g_str_equal (proto, "RTP/SAVPF"))
1394 stream->profile = GST_RTSP_PROFILE_SAVPF;
1398 len = gst_sdp_media_formats_len (media);
1399 for (i = 0; i < len; i++) {
1406 pt = atoi (gst_sdp_media_get_format (media, i));
1408 GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
1411 caps = gst_rtspsrc_media_to_caps (pt, media);
1413 GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
1417 /* do some tweaks */
1418 s = gst_caps_get_structure (caps, 0);
1419 if ((enc = gst_structure_get_string (s, "encoding-name"))) {
1420 stream->is_real = (strstr (enc, "-REAL") != NULL);
1421 if (strcmp (enc, "X-ASF-PF") == 0)
1422 stream->container = TRUE;
1424 GST_DEBUG ("mapping sdp session level attributes to caps");
1425 gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, caps);
1426 GST_DEBUG ("mapping sdp media level attributes to caps");
1427 gst_rtspsrc_sdp_attributes_to_caps (media->attributes, caps);
1429 /* the first pt will be the default */
1430 if (stream->ptmap->len == 0)
1431 stream->default_pt = pt;
1435 g_array_append_val (stream->ptmap, item);
1441 GST_ERROR_OBJECT (src, "can't find proto in media");
1446 GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
1451 static const gchar *
1452 get_aggregate_control (GstRTSPSrc * src)
1457 base = src->control;
1458 else if (src->content_base)
1459 base = src->content_base;
1460 else if (src->conninfo.url_str)
1461 base = src->conninfo.url_str;
1469 clear_ptmap_item (PtMapItem * item)
1472 gst_caps_unref (item->caps);
1475 static GstRTSPStream *
1476 gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
1478 GstRTSPStream *stream;
1479 const gchar *control_url;
1480 const GstSDPMedia *media;
1482 /* get media, should not return NULL */
1483 media = gst_sdp_message_get_media (sdp, idx);
1487 stream = g_new0 (GstRTSPStream, 1);
1488 stream->parent = src;
1489 /* we mark the pad as not linked, we will mark it as OK when we add the pad to
1491 stream->last_ret = GST_FLOW_NOT_LINKED;
1492 stream->added = FALSE;
1493 stream->setup = FALSE;
1494 stream->skipped = FALSE;
1496 stream->eos = FALSE;
1497 stream->discont = TRUE;
1498 stream->seqbase = -1;
1499 stream->timebase = -1;
1500 stream->send_ssrc = g_random_int ();
1501 stream->profile = GST_RTSP_PROFILE_AVP;
1502 stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
1503 g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
1505 /* collect bandwidth information for this steam. FIXME, configure in the RTP
1506 * session manager to scale RTCP. */
1507 gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
1509 /* collect connection info */
1510 gst_rtspsrc_collect_connections (src, sdp, media, stream);
1512 /* make the payload type map */
1513 gst_rtspsrc_collect_payloads (src, sdp, media, stream);
1515 /* collect port number */
1516 stream->port = gst_sdp_media_get_port (media);
1518 /* get control url to construct the setup url. The setup url is used to
1519 * configure the transport of the stream and is used to identity the stream in
1520 * the RTP-Info header field returned from PLAY. */
1521 control_url = gst_sdp_media_get_attribute_val (media, "control");
1522 if (control_url == NULL)
1523 control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
1525 GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
1526 GST_DEBUG_OBJECT (src, " port: %d", stream->port);
1527 GST_DEBUG_OBJECT (src, " container: %d", stream->container);
1528 GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
1530 if (control_url != NULL) {
1531 stream->control_url = g_strdup (control_url);
1532 /* Build a fully qualified url using the content_base if any or by prefixing
1533 * the original request.
1534 * If the control_url starts with a '/' or a non rtsp: protocol we will most
1535 * likely build a URL that the server will fail to understand, this is ok,
1536 * we will fail then. */
1537 if (g_str_has_prefix (control_url, "rtsp://"))
1538 stream->conninfo.location = g_strdup (control_url);
1543 if (g_strcmp0 (control_url, "*") == 0)
1546 base = get_aggregate_control (src);
1548 /* check if the base ends or control starts with / */
1549 has_slash = g_str_has_prefix (control_url, "/");
1550 has_slash = has_slash || g_str_has_suffix (base, "/");
1552 /* concatenate the two strings, insert / when not present */
1553 stream->conninfo.location =
1554 g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
1557 GST_DEBUG_OBJECT (src, " setup: %s",
1558 GST_STR_NULL (stream->conninfo.location));
1560 /* we keep track of all streams */
1561 src->streams = g_list_append (src->streams, stream);
1569 gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
1573 GST_DEBUG_OBJECT (src, "free stream %p", stream);
1575 g_array_free (stream->ptmap, TRUE);
1577 g_free (stream->destination);
1578 g_free (stream->control_url);
1579 g_free (stream->conninfo.location);
1581 for (i = 0; i < 2; i++) {
1582 if (stream->udpsrc[i]) {
1583 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
1584 gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
1585 gst_object_unref (stream->udpsrc[i]);
1587 if (stream->channelpad[i])
1588 gst_object_unref (stream->channelpad[i]);
1590 if (stream->udpsink[i]) {
1591 gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
1592 gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
1593 gst_object_unref (stream->udpsink[i]);
1596 if (stream->fakesrc) {
1597 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
1598 gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
1599 gst_object_unref (stream->fakesrc);
1601 if (stream->srcpad) {
1602 gst_pad_set_active (stream->srcpad, FALSE);
1604 gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
1606 if (stream->srtpenc)
1607 gst_object_unref (stream->srtpenc);
1608 if (stream->srtpdec)
1609 gst_object_unref (stream->srtpdec);
1610 if (stream->srtcpparams)
1611 gst_caps_unref (stream->srtcpparams);
1612 if (stream->rtcppad)
1613 gst_object_unref (stream->rtcppad);
1614 if (stream->session)
1615 g_object_unref (stream->session);
1616 if (stream->rtx_pt_map)
1617 gst_structure_free (stream->rtx_pt_map);
1622 gst_rtspsrc_cleanup (GstRTSPSrc * src)
1626 GST_DEBUG_OBJECT (src, "cleanup");
1628 for (walk = src->streams; walk; walk = g_list_next (walk)) {
1629 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
1631 gst_rtspsrc_stream_free (src, stream);
1633 g_list_free (src->streams);
1634 src->streams = NULL;
1636 if (src->manager_sig_id) {
1637 g_signal_handler_disconnect (src->manager, src->manager_sig_id);
1638 src->manager_sig_id = 0;
1640 gst_element_set_state (src->manager, GST_STATE_NULL);
1641 gst_bin_remove (GST_BIN_CAST (src), src->manager);
1642 src->manager = NULL;
1645 gst_structure_free (src->props);
1648 g_free (src->content_base);
1649 src->content_base = NULL;
1651 g_free (src->control);
1652 src->control = NULL;
1655 gst_rtsp_range_free (src->range);
1658 /* don't clear the SDP when it was used in the url */
1659 if (src->sdp && !src->from_sdp) {
1660 gst_sdp_message_free (src->sdp);
1664 src->need_segment = FALSE;
1666 if (src->provided_clock) {
1667 gst_object_unref (src->provided_clock);
1668 src->provided_clock = NULL;
1672 #define PARSE_INT(p, del, res) \
1675 p = strstr (p, del); \
1685 #define PARSE_STRING(p, del, res) \
1688 p = strstr (p, del); \
1700 #define SKIP_SPACES(p) \
1701 while (*p && g_ascii_isspace (*p)) \
1706 * <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1709 gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
1710 gint * rate, gchar ** params)
1714 p = (gchar *) rtpmap;
1716 PARSE_INT (p, " ", *payload);
1724 PARSE_STRING (p, "/", *name);
1725 if (*name == NULL) {
1726 GST_DEBUG ("no rate, name %s", p);
1727 /* no rate, assume -1 then, this is not supposed to happen but RealMedia
1728 * streams seem to omit the rate. */
1735 p = strstr (p, "/");
1753 parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
1755 gboolean res = FALSE;
1759 GstMIKEYMessage *msg;
1760 const GstMIKEYPayload *payload;
1761 const gchar *srtp_cipher;
1762 const gchar *srtp_auth;
1764 p = (gchar *) keymgmt;
1770 PARSE_STRING (p, " ", kmpid);
1771 if (!g_str_equal (kmpid, "mikey"))
1774 data = g_base64_decode (p, &size);
1778 msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
1783 srtp_cipher = "aes-128-icm";
1784 srtp_auth = "hmac-sha1-80";
1786 /* check the Security policy if any */
1787 if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
1788 GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
1791 if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
1794 len = gst_mikey_payload_sp_get_n_params (payload);
1795 for (i = 0; i < len; i++) {
1796 const GstMIKEYPayloadSPParam *param =
1797 gst_mikey_payload_sp_get_param (payload, i);
1799 switch (param->type) {
1800 case GST_MIKEY_SP_SRTP_ENC_ALG:
1801 switch (param->val[0]) {
1803 srtp_cipher = "null";
1807 srtp_cipher = "aes-128-icm";
1813 case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
1814 switch (param->val[0]) {
1815 case AES_128_KEY_LEN:
1816 srtp_cipher = "aes-128-icm";
1818 case AES_256_KEY_LEN:
1819 srtp_cipher = "aes-256-icm";
1825 case GST_MIKEY_SP_SRTP_AUTH_ALG:
1826 switch (param->val[0]) {
1832 srtp_auth = "hmac-sha1-80";
1838 case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
1839 switch (param->val[0]) {
1840 case HMAC_32_KEY_LEN:
1841 srtp_auth = "hmac-sha1-32";
1843 case HMAC_80_KEY_LEN:
1844 srtp_auth = "hmac-sha1-80";
1850 case GST_MIKEY_SP_SRTP_SRTP_ENC:
1852 case GST_MIKEY_SP_SRTP_SRTCP_ENC:
1860 if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
1863 GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
1864 const GstMIKEYPayload *sub;
1865 GstMIKEYPayloadKeyData *pkd;
1868 if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
1871 if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
1874 if (sub->type != GST_MIKEY_PT_KEY_DATA)
1877 pkd = (GstMIKEYPayloadKeyData *) sub;
1879 gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
1881 gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
1884 gst_caps_set_simple (caps,
1885 "srtp-cipher", G_TYPE_STRING, srtp_cipher,
1886 "srtp-auth", G_TYPE_STRING, srtp_auth,
1887 "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
1888 "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
1892 gst_mikey_message_unref (msg);
1898 * Mapping SDP attributes to caps
1900 * prepend 'a-' to IANA registered sdp attributes names
1901 * (ie: not prefixed with 'x-') in order to avoid
1902 * collision with gstreamer standard caps properties names
1905 gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
1907 if (attributes->len > 0) {
1911 s = gst_caps_get_structure (caps, 0);
1913 for (i = 0; i < attributes->len; i++) {
1914 GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
1915 gchar *tofree, *key;
1919 /* skip some of the attribute we already handle */
1920 if (!strcmp (key, "fmtp"))
1922 if (!strcmp (key, "rtpmap"))
1924 if (!strcmp (key, "control"))
1926 if (!strcmp (key, "range"))
1928 if (!strcmp (key, "framesize"))
1930 if (g_str_equal (key, "key-mgmt")) {
1931 parse_keymgmt (attr->value, caps);
1935 /* string must be valid UTF8 */
1936 if (!g_utf8_validate (attr->value, -1, NULL))
1939 if (!g_str_has_prefix (key, "x-"))
1940 tofree = key = g_strdup_printf ("a-%s", key);
1944 GST_DEBUG ("adding caps: %s=%s", key, attr->value);
1945 gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
1951 static const gchar *
1952 rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
1961 if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
1964 if (sscanf (attr, "%d ", &val) != 1)
1974 * Mapping of caps to and from SDP fields:
1976 * a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
1977 * a=framesize:<payload> <width>-<height>
1978 * a=fmtp:<payload> <param>[=<value>];...
1981 gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
1984 const gchar *rtpmap;
1986 const gchar *framesize;
1989 gchar *params = NULL;
1995 /* get and parse rtpmap */
1996 rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
1999 ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, ¶ms);
2001 g_warning ("error parsing rtpmap, ignoring");
2005 /* dynamic payloads need rtpmap or we fail */
2006 if (rtpmap == NULL && pt >= 96)
2009 /* check if we have a rate, if not, we need to look up the rate from the
2010 * default rates based on the payload types. */
2012 const GstRTPPayloadInfo *info;
2014 if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
2015 /* dynamic types, use media and encoding_name */
2016 tmp = g_ascii_strdown (media->media, -1);
2017 info = gst_rtp_payload_info_for_name (tmp, name);
2020 /* static types, use payload type */
2021 info = gst_rtp_payload_info_for_pt (pt);
2025 if ((rate = info->clock_rate) == 0)
2028 /* we fail if we cannot find one */
2033 tmp = g_ascii_strdown (media->media, -1);
2034 caps = gst_caps_new_simple ("application/x-unknown",
2035 "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
2037 s = gst_caps_get_structure (caps, 0);
2039 gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
2041 /* encoding name must be upper case */
2043 tmp = g_ascii_strup (name, -1);
2044 gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
2048 /* params must be lower case */
2049 if (params != NULL) {
2050 tmp = g_ascii_strdown (params, -1);
2051 gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
2055 /* parse optional fmtp: field */
2056 if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
2062 /* p is now of the format <payload> <param>[=<value>];... */
2063 PARSE_INT (p, " ", payload);
2064 if (payload != -1 && payload == pt) {
2068 /* <param>[=<value>] are separated with ';' */
2069 pairs = g_strsplit (p, ";", 0);
2070 for (i = 0; pairs[i]; i++) {
2072 const gchar *val, *key;
2074 /* the key may not have a '=', the value can have other '='s */
2075 valpos = strstr (pairs[i], "=");
2077 /* we have a '=' and thus a value, remove the '=' with \0 */
2079 /* value is everything between '=' and ';'. We split the pairs at ;
2080 * boundaries so we can take the remainder of the value. Some servers
2081 * put spaces around the value which we strip off here. Alternatively
2082 * we could strip those spaces in the depayloaders should these spaces
2083 * actually carry any meaning in the future. */
2084 val = g_strstrip (valpos + 1);
2086 /* simple <param>;.. is translated into <param>=1;... */
2089 /* strip the key of spaces, convert key to lowercase but not the value. */
2090 key = g_strstrip (pairs[i]);
2091 if (strlen (key) > 1) {
2092 tmp = g_ascii_strdown (key, -1);
2093 gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
2101 /* parse framesize: field */
2102 if ((framesize = gst_sdp_media_get_attribute_val (media, "framesize"))) {
2105 /* p is now of the format <payload> <width>-<height> */
2106 p = (gchar *) framesize;
2108 PARSE_INT (p, " ", payload);
2109 if (payload != -1 && payload == pt) {
2110 gst_structure_set (s, "a-framesize", G_TYPE_STRING, p, NULL);
2118 g_warning ("rtpmap type not given for dynamic payload %d", pt);
2123 g_warning ("rate unknown for payload type %d", pt);
2129 gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
2130 gint * rtpport, gint * rtcpport)
2133 GstStateChangeReturn ret;
2134 GstElement *udpsrc0, *udpsrc1;
2135 gint tmp_rtp, tmp_rtcp;
2139 src = stream->parent;
2145 /* Start at next port */
2146 tmp_rtp = src->next_port_num;
2148 if (stream->is_ipv6)
2149 host = "udp://[::0]";
2151 host = "udp://0.0.0.0";
2153 /* try to allocate 2 UDP ports, the RTP port should be an even
2154 * number and the RTCP port should be the next (uneven) port */
2157 if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
2158 tmp_rtp >= src->client_port_range.max)
2161 udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2162 if (udpsrc0 == NULL)
2163 goto no_udp_protocol;
2164 g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
2166 if (src->udp_buffer_size != 0)
2167 g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
2170 ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
2171 if (ret == GST_STATE_CHANGE_FAILURE) {
2173 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
2176 if (++count > src->retry)
2179 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2180 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2181 gst_object_unref (udpsrc0);
2184 GST_DEBUG_OBJECT (src, "retry %d", count);
2187 goto no_udp_protocol;
2190 g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
2191 GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
2193 /* check if port is even */
2194 if ((tmp_rtp & 0x01) != 0) {
2195 /* port not even, close and allocate another */
2196 if (++count > src->retry)
2199 GST_DEBUG_OBJECT (src, "RTP port not even");
2201 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2202 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2203 gst_object_unref (udpsrc0);
2206 GST_DEBUG_OBJECT (src, "retry %d", count);
2211 /* allocate port+1 for RTCP now */
2212 udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
2213 if (udpsrc1 == NULL)
2214 goto no_udp_rtcp_protocol;
2217 tmp_rtcp = tmp_rtp + 1;
2218 if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
2221 g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
2223 GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
2224 ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
2225 /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
2226 if (ret == GST_STATE_CHANGE_FAILURE) {
2227 GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
2229 if (++count > src->retry)
2232 GST_DEBUG_OBJECT (src, "free RTP udpsrc");
2233 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2234 gst_object_unref (udpsrc0);
2237 GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
2238 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2239 gst_object_unref (udpsrc1);
2243 GST_DEBUG_OBJECT (src, "retry %d", count);
2247 /* all fine, do port check */
2248 g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
2249 g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
2251 /* this should not happen... */
2252 if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
2255 /* we keep these elements, we configure all in configure_transport when the
2256 * server told us to really use the UDP ports. */
2257 stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
2258 stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
2259 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
2260 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
2262 /* keep track of next available port number when we have a range
2264 if (src->next_port_num != 0)
2265 src->next_port_num = tmp_rtcp + 1;
2272 GST_DEBUG_OBJECT (src, "could not get UDP source");
2277 GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
2281 no_udp_rtcp_protocol:
2283 GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
2288 GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
2289 tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
2295 gst_element_set_state (udpsrc0, GST_STATE_NULL);
2296 gst_object_unref (udpsrc0);
2299 gst_element_set_state (udpsrc1, GST_STATE_NULL);
2300 gst_object_unref (udpsrc1);
2307 gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
2312 gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
2314 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2315 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2318 for (i = 0; i < 2; i++) {
2319 if (stream->udpsrc[i])
2320 gst_element_set_state (stream->udpsrc[i], state);
2326 gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
2333 event = gst_event_new_flush_start ();
2334 GST_DEBUG_OBJECT (src, "start flush");
2336 state = GST_STATE_PAUSED;
2338 event = gst_event_new_flush_stop (FALSE);
2339 GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
2342 state = GST_STATE_PLAYING;
2344 state = GST_STATE_PAUSED;
2346 gst_rtspsrc_push_event (src, event);
2347 gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
2348 gst_rtspsrc_set_state (src, state);
2351 static GstRTSPResult
2352 gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
2353 GstRTSPMessage * message, GTimeVal * timeout)
2358 ret = gst_rtsp_connection_send (conn, message, timeout);
2360 ret = GST_RTSP_ERROR;
2365 static GstRTSPResult
2366 gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
2367 GstRTSPMessage * message, GTimeVal * timeout)
2372 ret = gst_rtsp_connection_receive (conn, message, timeout);
2374 ret = GST_RTSP_ERROR;
2380 gst_rtspsrc_get_position (GstRTSPSrc * src)
2385 query = gst_query_new_position (GST_FORMAT_TIME);
2386 /* should be known somewhere down the stream (e.g. jitterbuffer) */
2387 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2388 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2392 if (stream->srcpad) {
2393 if (gst_pad_query (stream->srcpad, query)) {
2394 gst_query_parse_position (query, &fmt, &pos);
2395 GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
2396 GST_TIME_ARGS (pos));
2397 src->last_pos = pos;
2407 gst_query_unref (query);
2411 gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
2416 GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
2418 gboolean flush, skip;
2421 GstSegment seeksegment = { 0, };
2425 GST_DEBUG_OBJECT (src, "doing seek with event");
2427 gst_event_parse_seek (event, &rate, &format, &flags,
2428 &cur_type, &cur, &stop_type, &stop);
2430 /* no negative rates yet */
2434 /* we need TIME format */
2435 if (format != src->segment.format)
2438 GST_DEBUG_OBJECT (src, "doing seek without event");
2440 cur_type = GST_SEEK_TYPE_SET;
2441 stop_type = GST_SEEK_TYPE_SET;
2444 /* get flush flag */
2445 flush = flags & GST_SEEK_FLAG_FLUSH;
2446 skip = flags & GST_SEEK_FLAG_SKIP;
2448 /* now we need to make sure the streaming thread is stopped. We do this by
2449 * either sending a FLUSH_START event downstream which will cause the
2450 * streaming thread to stop with a WRONG_STATE.
2451 * For a non-flushing seek we simply pause the task, which will happen as soon
2452 * as it completes one iteration (and thus might block when the sink is
2453 * blocking in preroll). */
2455 GST_DEBUG_OBJECT (src, "starting flush");
2456 gst_rtspsrc_flush (src, TRUE, FALSE);
2459 gst_task_pause (src->task);
2463 /* we should now be able to grab the streaming thread because we stopped it
2464 * with the above flush/pause code */
2465 GST_RTSP_STREAM_LOCK (src);
2467 GST_DEBUG_OBJECT (src, "stopped streaming");
2469 /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
2470 gst_rtspsrc_connection_flush (src, FALSE);
2472 /* copy segment, we need this because we still need the old
2473 * segment when we close the current segment. */
2474 memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
2476 /* configure the seek parameters in the seeksegment. We will then have the
2477 * right values in the segment to perform the seek */
2479 GST_DEBUG_OBJECT (src, "configuring seek");
2480 gst_segment_do_seek (&seeksegment, rate, format, flags,
2481 cur_type, cur, stop_type, stop, &update);
2484 /* figure out the last position we need to play. If it's configured (stop !=
2485 * -1), use that, else we play until the total duration of the file */
2486 if ((stop = seeksegment.stop) == -1)
2487 stop = seeksegment.duration;
2489 playing = (src->state == GST_RTSP_STATE_PLAYING);
2491 /* if we were playing, pause first */
2493 /* obtain current position in case seek fails */
2494 gst_rtspsrc_get_position (src);
2495 gst_rtspsrc_pause (src, FALSE);
2499 src->state = GST_RTSP_STATE_SEEKING;
2501 /* PLAY will add the range header now. */
2502 src->need_range = TRUE;
2504 /* and continue playing */
2506 gst_rtspsrc_play (src, &seeksegment, FALSE);
2508 /* prepare for streaming again */
2510 /* if we started flush, we stop now */
2511 GST_DEBUG_OBJECT (src, "stopping flush");
2512 gst_rtspsrc_flush (src, FALSE, playing);
2515 /* now we did the seek and can activate the new segment values */
2516 memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
2518 /* if we're doing a segment seek, post a SEGMENT_START message */
2519 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
2520 gst_element_post_message (GST_ELEMENT_CAST (src),
2521 gst_message_new_segment_start (GST_OBJECT_CAST (src),
2522 src->segment.format, src->segment.position));
2525 /* now create the newsegment */
2526 GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
2527 " to %" G_GINT64_FORMAT, src->segment.position, stop);
2530 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
2531 for (walk = src->streams; walk; walk = g_list_next (walk)) {
2532 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
2533 stream->discont = TRUE;
2536 GST_RTSP_STREAM_UNLOCK (src);
2543 GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
2548 GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
2554 gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
2558 gboolean res = TRUE;
2561 src = GST_RTSPSRC_CAST (parent);
2563 GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
2564 GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
2566 switch (GST_EVENT_TYPE (event)) {
2567 case GST_EVENT_SEEK:
2568 res = gst_rtspsrc_perform_seek (src, event);
2572 case GST_EVENT_NAVIGATION:
2573 case GST_EVENT_LATENCY:
2581 if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
2582 res = gst_pad_send_event (target, event);
2583 gst_object_unref (target);
2585 gst_event_unref (event);
2588 gst_event_unref (event);
2594 /* this is the final event function we receive on the internal source pad when
2595 * we deal with TCP connections */
2597 gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
2602 GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
2604 switch (GST_EVENT_TYPE (event)) {
2605 case GST_EVENT_SEEK:
2607 case GST_EVENT_NAVIGATION:
2608 case GST_EVENT_LATENCY:
2610 gst_event_unref (event);
2617 /* this is the final query function we receive on the internal source pad when
2618 * we deal with TCP connections */
2620 gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
2624 gboolean res = TRUE;
2626 src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
2628 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2629 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2631 switch (GST_QUERY_TYPE (query)) {
2632 case GST_QUERY_POSITION:
2637 case GST_QUERY_DURATION:
2641 gst_query_parse_duration (query, &format, NULL);
2644 case GST_FORMAT_TIME:
2645 gst_query_set_duration (query, format, src->segment.duration);
2653 case GST_QUERY_LATENCY:
2655 /* we are live with a min latency of 0 and unlimited max latency, this
2656 * result will be updated by the session manager if there is any. */
2657 gst_query_set_latency (query, TRUE, 0, -1);
2667 /* this query is executed on the ghost source pad exposed on rtspsrc. */
2669 gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
2673 gboolean res = FALSE;
2675 src = GST_RTSPSRC_CAST (parent);
2677 GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
2678 GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
2680 switch (GST_QUERY_TYPE (query)) {
2681 case GST_QUERY_DURATION:
2685 gst_query_parse_duration (query, &format, NULL);
2688 case GST_FORMAT_TIME:
2689 gst_query_set_duration (query, format, src->segment.duration);
2697 case GST_QUERY_SEEKING:
2701 gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
2702 if (format == GST_FORMAT_TIME) {
2704 src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
2706 /* seeking without duration is unlikely */
2707 seekable = seekable && src->seekable && src->segment.duration &&
2708 GST_CLOCK_TIME_IS_VALID (src->segment.duration);
2710 gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0,
2711 src->segment.duration);
2720 uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
2722 gst_query_set_uri (query, uri);
2730 GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
2732 /* forward the query to the proxy target pad */
2734 res = gst_pad_query (target, query);
2735 gst_object_unref (target);
2744 /* callback for RTCP messages to be sent to the server when operating in TCP
2746 static GstFlowReturn
2747 gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
2750 GstRTSPStream *stream;
2751 GstFlowReturn res = GST_FLOW_OK;
2756 GstRTSPMessage message = { 0 };
2757 GstRTSPConnection *conn;
2759 stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
2760 src = stream->parent;
2762 gst_buffer_map (buffer, &map, GST_MAP_READ);
2766 gst_rtsp_message_init_data (&message, stream->channel[1]);
2768 /* lend the body data to the message */
2769 gst_rtsp_message_take_body (&message, data, size);
2771 if (stream->conninfo.connection)
2772 conn = stream->conninfo.connection;
2774 conn = src->conninfo.connection;
2776 GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
2777 ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
2778 GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
2780 /* and steal it away again because we will free it when unreffing the
2782 gst_rtsp_message_steal_body (&message, &data, &size);
2783 gst_rtsp_message_unset (&message);
2785 gst_buffer_unmap (buffer, &map);
2786 gst_buffer_unref (buffer);
2791 static GstPadProbeReturn
2792 pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
2794 GstRTSPSrc *src = user_data;
2796 GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
2797 GST_DEBUG_PAD_NAME (pad));
2799 /* activate the streams */
2800 GST_OBJECT_LOCK (src);
2801 if (!src->need_activate)
2804 src->need_activate = FALSE;
2805 GST_OBJECT_UNLOCK (src);
2807 gst_rtspsrc_activate_streams (src);
2809 return GST_PAD_PROBE_OK;
2813 GST_OBJECT_UNLOCK (src);
2814 return GST_PAD_PROBE_OK;
2819 copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
2821 GstPad *gpad = GST_PAD_CAST (user_data);
2823 GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
2824 gst_pad_store_sticky_event (gpad, *event);
2829 /* this callback is called when the session manager generated a new src pad with
2830 * payloaded RTP packets. We simply ghost the pad here. */
2832 new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
2835 GstPadTemplate *template;
2838 GstRTSPStream *stream;
2841 GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
2843 GST_RTSP_STATE_LOCK (src);
2845 name = gst_object_get_name (GST_OBJECT_CAST (pad));
2846 if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
2847 goto unknown_stream;
2849 GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
2851 stream = find_stream (src, &id, (gpointer) find_stream_by_id);
2853 goto unknown_stream;
2856 stream->ssrc = ssrc;
2858 /* we'll add it later see below */
2859 stream->added = TRUE;
2861 /* check if we added all streams */
2863 for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
2864 GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
2866 GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
2867 ostream, ostream->container, ostream->added, ostream->setup);
2869 /* if we find a stream for which we did a setup that is not added, we
2870 * need to wait some more */
2871 if (ostream->setup && !ostream->added) {
2876 GST_RTSP_STATE_UNLOCK (src);
2878 /* create a new pad we will use to stream to */
2879 template = gst_static_pad_template_get (&rtptemplate);
2880 stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
2881 gst_object_unref (template);
2884 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
2885 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
2886 gst_pad_set_active (stream->srcpad, TRUE);
2887 gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
2888 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
2891 GST_DEBUG_OBJECT (src, "We added all streams");
2892 /* when we get here, all stream are added and we can fire the no-more-pads
2894 gst_element_no_more_pads (GST_ELEMENT_CAST (src));
2902 GST_DEBUG_OBJECT (src, "ignoring unknown stream");
2903 GST_RTSP_STATE_UNLOCK (src);
2910 stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
2914 len = stream->ptmap->len;
2915 for (i = 0; i < len; i++) {
2916 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
2924 request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
2926 GstRTSPStream *stream;
2929 GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
2931 GST_RTSP_STATE_LOCK (src);
2932 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
2934 goto unknown_stream;
2936 if ((caps = stream_get_caps_for_pt (stream, pt)))
2937 gst_caps_ref (caps);
2938 GST_RTSP_STATE_UNLOCK (src);
2944 GST_DEBUG_OBJECT (src, "unknown stream %d", session);
2945 GST_RTSP_STATE_UNLOCK (src);
2951 gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
2953 GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
2959 gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
2965 GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
2971 on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
2973 GstRTSPSrc *src = stream->parent;
2976 g_object_get (source, "ssrc", &ssrc, NULL);
2978 GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
2979 ssrc, stream->ssrc, stream->id);
2981 if (ssrc == stream->ssrc)
2982 gst_rtspsrc_do_stream_eos (src, stream);
2986 on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
2988 GstRTSPSrc *src = stream->parent;
2991 g_object_get (source, "ssrc", &ssrc, NULL);
2993 GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
2994 ssrc, stream->ssrc, stream->id);
2996 if (ssrc == stream->ssrc)
2997 gst_rtspsrc_do_stream_eos (src, stream);
3001 on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
3003 GstRTSPStream *stream;
3005 GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
3007 /* get stream for session */
3008 stream = find_stream (src, &session, (gpointer) find_stream_by_id);
3010 gst_rtspsrc_do_stream_eos (src, stream);
3015 on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
3017 GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
3022 set_manager_buffer_mode (GstRTSPSrc * src)
3024 GObjectClass *klass;
3026 if (src->manager == NULL)
3029 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3031 if (!g_object_class_find_property (klass, "buffer-mode"))
3034 if (src->buffer_mode != BUFFER_MODE_AUTO) {
3035 g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
3040 GST_DEBUG_OBJECT (src,
3041 "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
3043 if (src->provided_clock) {
3044 GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
3046 if (clock == src->provided_clock) {
3047 GST_DEBUG_OBJECT (src, "selected synced");
3048 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
3051 gst_object_unref (clock);
3056 /* Otherwise fall-through and use another buffer mode */
3058 gst_object_unref (clock);
3061 GST_DEBUG_OBJECT (src, "auto buffering mode");
3062 if (src->use_buffering) {
3063 GST_DEBUG_OBJECT (src, "selected buffer");
3064 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
3066 GST_DEBUG_OBJECT (src, "selected slave");
3067 g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
3072 request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
3074 GST_DEBUG ("request key %u", ssrc);
3075 return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
3079 request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
3081 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3082 if (stream->id != session)
3085 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3086 stream->profile != GST_RTSP_PROFILE_SAVPF)
3089 if (stream->srtpdec == NULL) {
3092 name = g_strdup_printf ("srtpdec_%u", session);
3093 stream->srtpdec = gst_element_factory_make ("srtpdec", name);
3096 g_signal_connect (stream->srtpdec, "request-key",
3097 (GCallback) request_key, stream);
3099 return gst_object_ref (stream->srtpdec);
3103 request_rtcp_encoder (GstElement * rtpbin, guint session,
3104 GstRTSPStream * stream)
3109 GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
3110 if (stream->id != session)
3113 if (stream->profile != GST_RTSP_PROFILE_SAVP &&
3114 stream->profile != GST_RTSP_PROFILE_SAVPF)
3117 if (stream->srtpenc == NULL) {
3120 name = g_strdup_printf ("srtpenc_%u", session);
3121 stream->srtpenc = gst_element_factory_make ("srtpenc", name);
3124 /* get RTCP crypto parameters from caps */
3125 s = gst_caps_get_structure (stream->srtcpparams, 0);
3129 GType ciphertype, authtype;
3130 GValue rtcp_cipher = G_VALUE_INIT, rtcp_auth = G_VALUE_INIT;
3132 ciphertype = g_type_from_name ("GstSrtpCipherType");
3133 authtype = g_type_from_name ("GstSrtpAuthType");
3134 g_value_init (&rtcp_cipher, ciphertype);
3135 g_value_init (&rtcp_auth, authtype);
3137 str = gst_structure_get_string (s, "srtcp-cipher");
3138 gst_value_deserialize (&rtcp_cipher, str);
3139 str = gst_structure_get_string (s, "srtcp-auth");
3140 gst_value_deserialize (&rtcp_auth, str);
3141 gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL);
3143 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-cipher",
3145 g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-auth",
3147 g_object_set (stream->srtpenc, "key", buf, NULL);
3149 g_value_unset (&rtcp_cipher);
3150 g_value_unset (&rtcp_auth);
3151 gst_buffer_unref (buf);
3154 name = g_strdup_printf ("rtcp_sink_%d", session);
3155 pad = gst_element_get_request_pad (stream->srtpenc, name);
3157 gst_object_unref (pad);
3159 return gst_object_ref (stream->srtpenc);
3163 request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPSrc * src)
3165 GstElement *rtx, *bin;
3168 GstRTSPStream *stream;
3170 stream = find_stream (src, &sessid, (gpointer) find_stream_by_id);
3172 GST_WARNING_OBJECT (src, "Stream %u not found", sessid);
3176 GST_INFO_OBJECT (src, "creating retransmision receiver for session %u "
3177 "with map %" GST_PTR_FORMAT, sessid, stream->rtx_pt_map);
3178 bin = gst_bin_new (NULL);
3179 rtx = gst_element_factory_make ("rtprtxreceive", NULL);
3180 g_object_set (rtx, "payload-type-map", stream->rtx_pt_map, NULL);
3181 gst_bin_add (GST_BIN (bin), rtx);
3183 pad = gst_element_get_static_pad (rtx, "src");
3184 name = g_strdup_printf ("src_%u", sessid);
3185 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3187 gst_object_unref (pad);
3189 pad = gst_element_get_static_pad (rtx, "sink");
3190 name = g_strdup_printf ("sink_%u", sessid);
3191 gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
3193 gst_object_unref (pad);
3199 add_retransmission (GstRTSPSrc * src, GstRTSPTransport * transport)
3203 gboolean do_retransmission = FALSE;
3205 if (transport->trans != GST_RTSP_TRANS_RTP)
3207 if (transport->profile != GST_RTSP_PROFILE_AVPF &&
3208 transport->profile != GST_RTSP_PROFILE_SAVPF)
3211 signal_id = g_signal_lookup ("request-aux-receiver",
3212 G_OBJECT_TYPE (src->manager));
3213 /* there's already something connected */
3214 if (g_signal_handler_find (src->manager, G_SIGNAL_MATCH_ID, signal_id, 0,
3215 NULL, NULL, NULL) != 0) {
3216 GST_DEBUG_OBJECT (src, "Not adding RTX AUX element as "
3217 "\"request-aux-receiver\" signal is "
3218 "already used by the application");
3222 /* build the retransmission payload type map */
3223 for (walk = src->streams; walk; walk = g_list_next (walk)) {
3224 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
3225 gboolean do_retransmission_stream = FALSE;
3228 if (stream->rtx_pt_map)
3229 gst_structure_free (stream->rtx_pt_map);
3230 stream->rtx_pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
3232 for (i = 0; i < stream->ptmap->len; i++) {
3233 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
3234 GstStructure *s = gst_caps_get_structure (item->caps, 0);
3235 const gchar *encoding;
3237 /* we only care about RTX streams */
3238 if ((encoding = gst_structure_get_string (s, "encoding-name"))
3239 && g_strcmp0 (encoding, "RTX") == 0) {
3240 const gchar *stream_pt_s;
3243 if (gst_structure_get_int (s, "payload", &rtx_pt)
3244 && (stream_pt_s = gst_structure_get_string (s, "apt"))) {
3247 gst_structure_set (stream->rtx_pt_map, stream_pt_s, G_TYPE_UINT,
3249 do_retransmission_stream = TRUE;
3255 if (do_retransmission_stream) {
3256 GST_DEBUG_OBJECT (src, "built retransmission payload map for stream "
3257 "id %i: %" GST_PTR_FORMAT, stream->id, stream->rtx_pt_map);
3258 do_retransmission = TRUE;
3260 GST_DEBUG_OBJECT (src, "no retransmission payload map for stream "
3261 "id %i", stream->id);
3262 gst_structure_free (stream->rtx_pt_map);
3263 stream->rtx_pt_map = NULL;
3267 if (do_retransmission) {
3268 GST_DEBUG_OBJECT (src, "Enabling retransmissions");
3270 g_object_set (src->manager, "do-retransmission", TRUE, NULL);
3272 /* enable RFC4588 retransmission handling by setting rtprtxreceive
3273 * as the "aux" element of rtpbin */
3274 g_signal_connect (src->manager, "request-aux-receiver",
3275 (GCallback) request_aux_receiver, src);
3277 GST_DEBUG_OBJECT (src,
3278 "Not enabling retransmissions as no stream had a retransmission payload map");
3282 /* try to get and configure a manager */
3284 gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
3285 GstRTSPTransport * transport)
3287 const gchar *manager;
3289 GstStateChangeReturn ret;
3291 /* find a manager */
3292 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
3296 GST_DEBUG_OBJECT (src, "using manager %s", manager);
3298 /* configure the manager */
3299 if (src->manager == NULL) {
3300 GObjectClass *klass;
3302 if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
3304 if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
3308 goto use_no_manager;
3310 if (!(src->manager = gst_element_factory_make (manager, "manager")))
3311 goto manager_failed;
3314 /* we manage this element */
3315 gst_element_set_locked_state (src->manager, TRUE);
3316 gst_bin_add (GST_BIN_CAST (src), src->manager);
3318 ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
3319 if (ret == GST_STATE_CHANGE_FAILURE)
3320 goto start_manager_failure;
3322 g_object_set (src->manager, "latency", src->latency, NULL);
3324 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
3326 if (g_object_class_find_property (klass, "ntp-sync")) {
3327 g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
3330 if (g_object_class_find_property (klass, "use-pipeline-clock")) {
3331 g_object_set (src->manager, "use-pipeline-clock",
3332 src->use_pipeline_clock, NULL);
3335 if (src->sdes && g_object_class_find_property (klass, "sdes")) {
3336 g_object_set (src->manager, "sdes", src->sdes, NULL);
3339 if (g_object_class_find_property (klass, "drop-on-latency")) {
3340 g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
3344 /* buffer mode pauses are handled by adding offsets to buffer times,
3345 * but some depayloaders may have a hard time syncing output times
3346 * with such input times, e.g. container ones, most notably ASF */
3347 /* TODO alternatives are having an event that indicates these shifts,
3348 * or having rtsp extensions provide suggestion on buffer mode */
3349 /* valid duration implies not likely live pipeline,
3350 * so slaving in jitterbuffer does not make much sense
3351 * (and might mess things up due to bursts) */
3352 if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
3353 src->segment.duration && stream->container) {
3354 src->use_buffering = TRUE;
3356 src->use_buffering = FALSE;
3359 set_manager_buffer_mode (src);
3361 /* connect to signals */
3362 GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
3364 src->manager_sig_id =
3365 g_signal_connect (src->manager, "pad-added",
3366 (GCallback) new_manager_pad, src);
3367 src->manager_ptmap_id =
3368 g_signal_connect (src->manager, "request-pt-map",
3369 (GCallback) request_pt_map, src);
3371 g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
3374 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
3377 if (src->do_retransmission)
3378 add_retransmission (src, transport);
3380 g_signal_connect (src->manager, "request-rtp-decoder",
3381 (GCallback) request_rtp_decoder, stream);
3382 g_signal_connect (src->manager, "request-rtcp-decoder",
3383 (GCallback) request_rtp_decoder, stream);
3384 g_signal_connect (src->manager, "request-rtcp-encoder",
3385 (GCallback) request_rtcp_encoder, stream);
3387 /* we stream directly to the manager, get some pads. Each RTSP stream goes
3388 * into a separate RTP session. */
3389 name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
3390 stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
3392 name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
3393 stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
3396 /* now configure the bandwidth in the manager */
3397 if (g_signal_lookup ("get-internal-session",
3398 G_OBJECT_TYPE (src->manager)) != 0) {
3399 GObject *rtpsession;
3401 g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
3404 GstRTPProfile rtp_profile;
3406 GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
3408 stream->session = rtpsession;
3410 if (stream->as_bandwidth != -1) {
3411 GST_INFO_OBJECT (src, "setting AS: %f",
3412 (gdouble) (stream->as_bandwidth * 1000));
3413 g_object_set (rtpsession, "bandwidth",
3414 (gdouble) (stream->as_bandwidth * 1000), NULL);
3416 if (stream->rr_bandwidth != -1) {
3417 GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
3418 g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
3421 if (stream->rs_bandwidth != -1) {
3422 GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
3423 g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
3427 switch (stream->profile) {
3428 case GST_RTSP_PROFILE_AVPF:
3429 rtp_profile = GST_RTP_PROFILE_AVPF;
3431 case GST_RTSP_PROFILE_SAVP:
3432 rtp_profile = GST_RTP_PROFILE_SAVP;
3434 case GST_RTSP_PROFILE_SAVPF:
3435 rtp_profile = GST_RTP_PROFILE_SAVPF;
3437 case GST_RTSP_PROFILE_AVP:
3439 rtp_profile = GST_RTP_PROFILE_AVP;
3443 g_object_set (rtpsession, "rtp-profile", rtp_profile, NULL);
3445 g_object_set (rtpsession, "probation", src->probation, NULL);
3447 g_object_set (rtpsession, "internal-ssrc", stream->send_ssrc, NULL);
3449 g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
3451 g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
3453 g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
3455 g_signal_connect (rtpsession, "on-ssrc-active",
3456 (GCallback) on_ssrc_active, stream);
3467 GST_DEBUG_OBJECT (src, "cannot get a session manager");
3472 GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
3475 start_manager_failure:
3477 GST_DEBUG_OBJECT (src, "could not start session manager");
3482 /* free the UDP sources allocated when negotiating a transport.
3483 * This function is called when the server negotiated to a transport where the
3484 * UDP sources are not needed anymore, such as TCP or multicast. */
3486 gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
3490 for (i = 0; i < 2; i++) {
3491 if (stream->udpsrc[i]) {
3492 GST_DEBUG ("free UDP source %d for stream %p", i, stream);
3493 gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
3494 gst_object_unref (stream->udpsrc[i]);
3495 stream->udpsrc[i] = NULL;
3500 /* for TCP, create pads to send and receive data to and from the manager and to
3501 * intercept various events and queries
3504 gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
3505 GstRTSPTransport * transport, GstPad ** outpad)
3508 GstPadTemplate *template;
3509 GstPad *pad0, *pad1;
3511 /* configure for interleaved delivery, nothing needs to be done
3512 * here, the loop function will call the chain functions of the
3513 * session manager. */
3514 stream->channel[0] = transport->interleaved.min;
3515 stream->channel[1] = transport->interleaved.max;
3516 GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
3517 stream->channel[0], stream->channel[1]);
3519 /* we can remove the allocated UDP ports now */
3520 gst_rtspsrc_stream_free_udp (stream);
3522 /* no session manager, send data to srcpad directly */
3523 if (!stream->channelpad[0]) {
3524 GST_DEBUG_OBJECT (src, "no manager, creating pad");
3526 /* create a new pad we will use to stream to */
3527 name = g_strdup_printf ("stream_%u", stream->id);
3528 template = gst_static_pad_template_get (&rtptemplate);
3529 stream->channelpad[0] = gst_pad_new_from_template (template, name);
3530 gst_object_unref (template);
3533 /* set caps and activate */
3534 gst_pad_use_fixed_caps (stream->channelpad[0]);
3535 gst_pad_set_active (stream->channelpad[0], TRUE);
3537 *outpad = gst_object_ref (stream->channelpad[0]);
3539 GST_DEBUG_OBJECT (src, "using manager source pad");
3541 template = gst_static_pad_template_get (&anysrctemplate);
3543 /* allocate pads for sending the channel data into the manager */
3544 pad0 = gst_pad_new_from_template (template, "internalsrc_0");
3545 gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
3546 gst_object_unref (stream->channelpad[0]);
3547 stream->channelpad[0] = pad0;
3548 gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
3549 gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
3550 gst_pad_set_element_private (pad0, src);
3551 gst_pad_set_active (pad0, TRUE);
3553 if (stream->channelpad[1]) {
3554 /* if we have a sinkpad for the other channel, create a pad and link to the
3556 pad1 = gst_pad_new_from_template (template, "internalsrc_1");
3557 gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
3558 gst_pad_link_full (pad1, stream->channelpad[1],
3559 GST_PAD_LINK_CHECK_NOTHING);
3560 gst_object_unref (stream->channelpad[1]);
3561 stream->channelpad[1] = pad1;
3562 gst_pad_set_active (pad1, TRUE);
3564 gst_object_unref (template);
3566 /* setup RTCP transport back to the server if we have to. */
3567 if (src->manager && src->do_rtcp) {
3570 template = gst_static_pad_template_get (&anysinktemplate);
3572 stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
3573 gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
3574 gst_pad_set_element_private (stream->rtcppad, stream);
3575 gst_pad_set_active (stream->rtcppad, TRUE);
3577 /* get session RTCP pad */
3578 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3579 pad = gst_element_get_request_pad (src->manager, name);
3584 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3585 gst_object_unref (pad);
3588 gst_object_unref (template);
3594 gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
3595 GstRTSPTransport * transport, const gchar ** destination, gint * min,
3596 gint * max, guint * ttl)
3598 if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
3600 if (!(*destination = transport->destination))
3601 *destination = stream->destination;
3604 /* transport first */
3605 *min = transport->port.min;
3606 *max = transport->port.max;
3607 if (*min == -1 && *max == -1) {
3608 /* then try from SDP */
3609 if (stream->port != 0) {
3610 *min = stream->port;
3611 *max = stream->port + 1;
3617 if (!(*ttl = transport->ttl))
3622 /* first take the source, then the endpoint to figure out where to send
3624 if (!(*destination = transport->source)) {
3625 if (src->conninfo.connection)
3626 *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
3627 else if (stream->conninfo.connection)
3629 gst_rtsp_connection_get_ip (stream->conninfo.connection);
3633 /* for unicast we only expect the ports here */
3634 *min = transport->server_port.min;
3635 *max = transport->server_port.max;
3640 /* For multicast create UDP sources and join the multicast group. */
3642 gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
3643 GstRTSPTransport * transport, GstPad ** outpad)
3646 const gchar *destination;
3649 GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
3651 /* we can remove the allocated UDP ports now */
3652 gst_rtspsrc_stream_free_udp (stream);
3654 gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
3657 /* we need a destination now */
3658 if (destination == NULL)
3659 goto no_destination;
3661 /* we really need ports now or we won't be able to receive anything at all */
3662 if (min == -1 && max == -1)
3665 GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
3666 destination, min, max);
3668 /* creating UDP source for RTP */
3670 uri = g_strdup_printf ("udp://%s:%d", destination, min);
3672 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3674 if (stream->udpsrc[0] == NULL)
3677 /* take ownership */
3678 gst_object_ref_sink (stream->udpsrc[0]);
3680 if (src->udp_buffer_size != 0)
3681 g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
3682 src->udp_buffer_size, NULL);
3684 if (src->multi_iface != NULL)
3685 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3686 src->multi_iface, NULL);
3689 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3690 gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
3693 /* creating another UDP source for RTCP */
3697 uri = g_strdup_printf ("udp://%s:%d", destination, max);
3699 gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
3701 if (stream->udpsrc[1] == NULL)
3704 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3705 stream->profile == GST_RTSP_PROFILE_SAVPF)
3706 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3708 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3709 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3710 gst_caps_unref (caps);
3712 /* take ownership */
3713 gst_object_ref_sink (stream->udpsrc[1]);
3715 if (src->multi_iface != NULL)
3716 g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
3717 src->multi_iface, NULL);
3719 gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
3726 GST_DEBUG_OBJECT (src, "no UDP source element found");
3731 GST_DEBUG_OBJECT (src, "no destination found");
3736 GST_DEBUG_OBJECT (src, "no ports found");
3741 /* configure the remainder of the UDP ports */
3743 gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
3744 GstRTSPTransport * transport, GstPad ** outpad)
3746 /* we manage the UDP elements now. For unicast, the UDP sources where
3747 * allocated in the stream when we suggested a transport. */
3748 if (stream->udpsrc[0]) {
3751 gst_element_set_locked_state (stream->udpsrc[0], TRUE);
3752 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
3754 GST_DEBUG_OBJECT (src, "setting up UDP source");
3756 /* configure a timeout on the UDP port. When the timeout message is
3757 * posted, we assume UDP transport is not possible. We reconnect using TCP
3759 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
3760 src->udp_timeout * 1000, NULL);
3762 if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
3763 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
3765 /* get output pad of the UDP source. */
3766 *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
3768 /* save it so we can unblock */
3769 stream->blockedpad = *outpad;
3771 /* configure pad block on the pad. As soon as there is dataflow on the
3772 * UDP source, we know that UDP is not blocked by a firewall and we can
3773 * configure all the streams to let the application autoplug decoders. */
3775 gst_pad_add_probe (stream->blockedpad,
3776 GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
3777 GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
3779 if (stream->channelpad[0]) {
3780 GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
3781 /* configure for UDP delivery, we need to connect the UDP pads to
3782 * the session plugin. */
3783 gst_pad_link_full (*outpad, stream->channelpad[0],
3784 GST_PAD_LINK_CHECK_NOTHING);
3785 gst_object_unref (*outpad);
3787 /* we connected to pad-added signal to get pads from the manager */
3789 GST_DEBUG_OBJECT (src, "using UDP src pad as output");
3794 if (stream->udpsrc[1]) {
3797 gst_element_set_locked_state (stream->udpsrc[1], TRUE);
3798 gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
3800 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
3801 stream->profile == GST_RTSP_PROFILE_SAVPF)
3802 caps = gst_caps_new_empty_simple ("application/x-srtcp");
3804 caps = gst_caps_new_empty_simple ("application/x-rtcp");
3805 g_object_set (stream->udpsrc[1], "caps", caps, NULL);
3806 gst_caps_unref (caps);
3808 if (stream->channelpad[1]) {
3811 GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
3813 pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
3814 gst_pad_link_full (pad, stream->channelpad[1],
3815 GST_PAD_LINK_CHECK_NOTHING);
3816 gst_object_unref (pad);
3818 /* leave unlinked */
3824 /* configure the UDP sink back to the server for status reports */
3826 gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
3827 GstRTSPStream * stream, GstRTSPTransport * transport)
3830 gint rtp_port, rtcp_port;
3831 gboolean do_rtp, do_rtcp;
3832 const gchar *destination;
3837 /* get transport info */
3838 gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
3839 &rtp_port, &rtcp_port, &ttl);
3841 /* see what we need to do */
3842 do_rtp = (rtp_port != -1);
3843 /* it's possible that the server does not want us to send RTCP in which case
3845 do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
3847 /* we need a destination when we have RTP or RTCP ports */
3848 if (destination == NULL && (do_rtp || do_rtcp))
3849 goto no_destination;
3851 /* try to construct the fakesrc to the RTP port of the server to open up any
3854 GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
3857 uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
3858 stream->udpsink[0] =
3859 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3861 if (stream->udpsink[0] == NULL)
3862 goto no_sink_element;
3864 /* don't join multicast group, we will have the source socket do that */
3865 /* no sync or async state changes needed */
3866 g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
3867 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3869 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3871 if (stream->udpsrc[0]) {
3872 /* configure socket, we give it the same UDP socket as the udpsrc for RTP
3873 * so that NAT firewalls will open a hole for us */
3874 g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
3875 GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
3876 /* configure socket and make sure udpsink does not close it when shutting
3877 * down, it belongs to udpsrc after all. */
3878 g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
3879 "close-socket", FALSE, NULL);
3880 g_object_unref (socket);
3883 /* the source for the dummy packets to open up NAT */
3884 stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
3885 if (stream->fakesrc == NULL)
3886 goto no_fakesrc_element;
3888 /* random data in 5 buffers, a size of 200 bytes should be fine */
3889 g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
3890 "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
3892 /* we don't want to consider this a sink */
3893 GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
3895 /* keep everything locked */
3896 gst_element_set_locked_state (stream->udpsink[0], TRUE);
3897 gst_element_set_locked_state (stream->fakesrc, TRUE);
3899 gst_object_ref (stream->udpsink[0]);
3900 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
3901 gst_object_ref (stream->fakesrc);
3902 gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
3904 gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
3905 "sink", GST_PAD_LINK_CHECK_NOTHING);
3908 GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
3911 uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
3912 stream->udpsink[1] =
3913 gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
3915 if (stream->udpsink[1] == NULL)
3916 goto no_sink_element;
3918 /* don't join multicast group, we will have the source socket do that */
3919 /* no sync or async state changes needed */
3920 g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
3921 "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
3923 g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
3925 if (stream->udpsrc[1]) {
3926 /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
3927 * because some servers check the port number of where it sends RTCP to identify
3928 * the RTCP packets it receives */
3929 g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
3930 GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
3931 /* configure socket and make sure udpsink does not close it when shutting
3932 * down, it belongs to udpsrc after all. */
3933 g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
3934 "close-socket", FALSE, NULL);
3935 g_object_unref (socket);
3938 /* we don't want to consider this a sink */
3939 GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
3941 /* we keep this playing always */
3942 gst_element_set_locked_state (stream->udpsink[1], TRUE);
3943 gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
3945 gst_object_ref (stream->udpsink[1]);
3946 gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
3948 stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
3950 /* get session RTCP pad */
3951 name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
3952 pad = gst_element_get_request_pad (src->manager, name);
3957 gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
3958 gst_object_unref (pad);
3967 GST_DEBUG_OBJECT (src, "no destination address specified");
3972 GST_DEBUG_OBJECT (src, "no UDP sink element found");
3977 GST_DEBUG_OBJECT (src, "no fakesrc element found");
3982 /* sets up all elements needed for streaming over the specified transport.
3983 * Does not yet expose the element pads, this will be done when there is actuall
3984 * dataflow detected, which might never happen when UDP is blocked in a
3985 * firewall, for example.
3988 gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
3989 GstRTSPTransport * transport)
3992 GstPad *outpad = NULL;
3993 GstPadTemplate *template;
3995 const gchar *media_type;
3998 src = stream->parent;
4000 GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
4002 /* get the proper media type for this stream now */
4003 if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
4004 goto unknown_transport;
4006 goto unknown_transport;
4008 /* configure the final media type */
4009 GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
4011 len = stream->ptmap->len;
4012 for (i = 0; i < len; i++) {
4014 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
4016 if (item->caps == NULL)
4019 s = gst_caps_get_structure (item->caps, 0);
4020 gst_structure_set_name (s, media_type);
4021 /* set ssrc if known */
4022 if (transport->ssrc)
4023 gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
4026 /* try to get and configure a manager, channelpad[0-1] will be configured with
4027 * the pads for the manager, or NULL when no manager is needed. */
4028 if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
4031 switch (transport->lower_transport) {
4032 case GST_RTSP_LOWER_TRANS_TCP:
4033 if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
4034 goto transport_failed;
4036 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
4037 if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
4038 goto transport_failed;
4039 /* fallthrough, the rest is the same for UDP and MCAST */
4040 case GST_RTSP_LOWER_TRANS_UDP:
4041 if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
4042 goto transport_failed;
4043 /* configure udpsinks back to the server for RTCP messages and for the
4044 * dummy RTP messages to open NAT. */
4045 if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
4046 goto transport_failed;
4049 goto unknown_transport;
4053 GST_DEBUG_OBJECT (src, "creating ghostpad");
4055 gst_pad_use_fixed_caps (outpad);
4057 /* create ghostpad, don't add just yet, this will be done when we activate
4059 name = g_strdup_printf ("stream_%u", stream->id);
4060 template = gst_static_pad_template_get (&rtptemplate);
4061 stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
4062 gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
4063 gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
4064 gst_object_unref (template);
4067 gst_object_unref (outpad);
4069 /* mark pad as ok */
4070 stream->last_ret = GST_FLOW_OK;
4077 GST_DEBUG_OBJECT (src, "failed to configure transport");
4082 GST_DEBUG_OBJECT (src, "unknown transport");
4087 GST_DEBUG_OBJECT (src, "cannot get a session manager");
4092 /* send a couple of dummy random packets on the receiver RTP port to the server,
4093 * this should make a firewall think we initiated the data transfer and
4094 * hopefully allow packets to go from the sender port to our RTP receiver port */
4096 gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
4100 if (src->nat_method != GST_RTSP_NAT_DUMMY)
4103 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4104 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4106 if (stream->fakesrc && stream->udpsink[0]) {
4107 GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
4108 gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
4109 gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
4110 gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
4111 gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
4117 /* Adds the source pads of all configured streams to the element.
4118 * This code is performed when we detected dataflow.
4120 * We detect dataflow from either the _loop function or with pad probes on the
4124 gst_rtspsrc_activate_streams (GstRTSPSrc * src)
4128 GST_DEBUG_OBJECT (src, "activating streams");
4130 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4131 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4133 if (stream->udpsrc[0]) {
4134 /* remove timeout, we are streaming now and timeouts will be handled by
4135 * the session manager and jitter buffer */
4136 g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
4138 if (stream->srcpad) {
4139 GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
4140 gst_pad_set_active (stream->srcpad, TRUE);
4142 /* if we don't have a session manager, set the caps now. If we have a
4143 * session, we will get a notification of the pad and the caps. */
4144 if (!src->manager) {
4147 caps = stream_get_caps_for_pt (stream, stream->default_pt);
4148 GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
4149 gst_pad_set_caps (stream->srcpad, caps);
4152 if (!stream->added) {
4153 GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
4154 gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
4155 stream->added = TRUE;
4160 /* unblock all pads */
4161 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4162 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4164 if (stream->blockid) {
4165 GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
4166 gst_pad_remove_probe (stream->blockedpad, stream->blockid);
4167 stream->blockid = 0;
4175 gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
4176 gboolean reset_manager)
4179 guint64 start, stop;
4180 gdouble play_speed, play_scale;
4182 GST_DEBUG_OBJECT (src, "configuring stream caps");
4184 start = segment->position;
4185 stop = segment->duration;
4186 play_speed = segment->rate;
4187 play_scale = segment->applied_rate;
4189 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4190 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4196 len = stream->ptmap->len;
4197 for (j = 0; j < len; j++) {
4199 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
4201 if (item->caps == NULL)
4204 caps = gst_caps_make_writable (item->caps);
4206 if (stream->timebase != -1)
4207 gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
4208 (guint) stream->timebase, NULL);
4209 if (stream->seqbase != -1)
4210 gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
4211 (guint) stream->seqbase, NULL);
4212 gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
4214 gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
4215 gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
4216 gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
4219 GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
4222 if (item->pt == stream->default_pt && stream->udpsrc[0]) {
4223 g_object_set (stream->udpsrc[0], "caps", caps, NULL);
4227 if (reset_manager && src->manager) {
4228 GST_DEBUG_OBJECT (src, "clear session");
4229 g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
4233 static GstFlowReturn
4234 gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
4239 /* store the value */
4240 stream->last_ret = ret;
4242 /* if it's success we can return the value right away */
4243 if (ret == GST_FLOW_OK)
4246 /* any other error that is not-linked can be returned right
4248 if (ret != GST_FLOW_NOT_LINKED)
4251 /* only return NOT_LINKED if all other pads returned NOT_LINKED */
4252 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4253 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4255 ret = ostream->last_ret;
4256 /* some other return value (must be SUCCESS but we can return
4257 * other values as well) */
4258 if (ret != GST_FLOW_NOT_LINKED)
4261 /* if we get here, all other pads were unlinked and we return
4262 * NOT_LINKED then */
4268 gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
4271 gboolean res = TRUE;
4273 /* only streams that have a connection to the outside world */
4277 if (stream->udpsrc[0]) {
4278 gst_event_ref (event);
4279 res = gst_element_send_event (stream->udpsrc[0], event);
4280 } else if (stream->channelpad[0]) {
4281 gst_event_ref (event);
4282 if (GST_PAD_IS_SRC (stream->channelpad[0]))
4283 res = gst_pad_push_event (stream->channelpad[0], event);
4285 res = gst_pad_send_event (stream->channelpad[0], event);
4288 if (stream->udpsrc[1]) {
4289 gst_event_ref (event);
4290 res &= gst_element_send_event (stream->udpsrc[1], event);
4291 } else if (stream->channelpad[1]) {
4292 gst_event_ref (event);
4293 if (GST_PAD_IS_SRC (stream->channelpad[1]))
4294 res &= gst_pad_push_event (stream->channelpad[1], event);
4296 res &= gst_pad_send_event (stream->channelpad[1], event);
4300 gst_event_unref (event);
4306 gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
4309 gboolean res = TRUE;
4311 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4312 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4314 gst_event_ref (event);
4315 res &= gst_rtspsrc_stream_push_event (src, ostream, event);
4317 gst_event_unref (event);
4322 static GstRTSPResult
4323 gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4328 if (info->connection == NULL) {
4329 if (info->url == NULL) {
4330 GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
4331 if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
4335 /* create connection */
4336 GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
4337 if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
4338 goto could_not_create;
4341 g_free (info->url_str);
4342 info->url_str = gst_rtsp_url_get_request_uri (info->url);
4344 GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
4346 if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
4347 if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
4348 src->tls_validation_flags))
4349 GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
4351 if (src->tls_database)
4352 gst_rtsp_connection_set_tls_database (info->connection,
4356 if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
4357 gst_rtsp_connection_set_tunneled (info->connection, TRUE);
4359 if (src->proxy_host) {
4360 GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
4362 gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
4367 if (!info->connected) {
4370 GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
4371 ("Connecting to %s", info->location));
4372 GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
4374 gst_rtsp_connection_connect (info->connection,
4375 src->ptcp_timeout)) < 0)
4376 goto could_not_connect;
4378 info->connected = TRUE;
4385 GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
4390 gchar *str = gst_rtsp_strresult (res);
4391 GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
4397 gchar *str = gst_rtsp_strresult (res);
4398 GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
4404 static GstRTSPResult
4405 gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
4408 GST_RTSP_STATE_LOCK (src);
4409 if (info->connected) {
4410 GST_DEBUG_OBJECT (src, "closing connection...");
4411 gst_rtsp_connection_close (info->connection);
4412 info->connected = FALSE;
4414 if (free && info->connection) {
4415 /* free connection */
4416 GST_DEBUG_OBJECT (src, "freeing connection...");
4417 gst_rtsp_connection_free (info->connection);
4418 info->connection = NULL;
4420 GST_RTSP_STATE_UNLOCK (src);
4424 static GstRTSPResult
4425 gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
4430 GST_DEBUG_OBJECT (src, "reconnecting connection...");
4431 gst_rtsp_conninfo_close (src, info, FALSE);
4432 res = gst_rtsp_conninfo_connect (src, info, async);
4438 gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
4442 GST_DEBUG_OBJECT (src, "set flushing %d", flush);
4443 GST_RTSP_STATE_LOCK (src);
4444 if (src->conninfo.connection && src->conninfo.flushing != flush) {
4445 GST_DEBUG_OBJECT (src, "connection flush");
4446 gst_rtsp_connection_flush (src->conninfo.connection, flush);
4447 src->conninfo.flushing = flush;
4449 for (walk = src->streams; walk; walk = g_list_next (walk)) {
4450 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
4451 if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
4452 GST_DEBUG_OBJECT (src, "stream %p flush", stream);
4453 gst_rtsp_connection_flush (stream->conninfo.connection, flush);
4454 stream->conninfo.flushing = flush;
4457 GST_RTSP_STATE_UNLOCK (src);
4460 /* FIXME, handle server request, reply with OK, for now */
4461 static GstRTSPResult
4462 gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
4463 GstRTSPMessage * request)
4465 GstRTSPMessage response = { 0 };
4468 GST_DEBUG_OBJECT (src, "got server request message");
4471 gst_rtsp_message_dump (request);
4473 res = gst_rtsp_ext_list_receive_request (src->extensions, request);
4475 if (res == GST_RTSP_ENOTIMPL) {
4476 /* default implementation, send OK */
4477 GST_DEBUG_OBJECT (src, "prepare OK reply");
4479 gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
4484 /* let app parse and reply */
4485 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
4486 0, request, &response);
4489 gst_rtsp_message_dump (&response);
4491 res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
4495 gst_rtsp_message_unset (&response);
4496 } else if (res == GST_RTSP_EEOF)
4504 gst_rtsp_message_unset (&response);
4509 /* send server keep-alive */
4510 static GstRTSPResult
4511 gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
4513 GstRTSPMessage request = { 0 };
4515 GstRTSPMethod method;
4516 const gchar *control;
4518 if (src->do_rtsp_keep_alive == FALSE) {
4519 GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
4520 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4524 GST_DEBUG_OBJECT (src, "creating server keep-alive");
4526 /* find a method to use for keep-alive */
4527 if (src->methods & GST_RTSP_GET_PARAMETER)
4528 method = GST_RTSP_GET_PARAMETER;
4530 method = GST_RTSP_OPTIONS;
4532 control = get_aggregate_control (src);
4533 if (control == NULL)
4536 res = gst_rtsp_message_init_request (&request, method, control);
4541 gst_rtsp_message_dump (&request);
4544 gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
4549 gst_rtsp_connection_reset_timeout (src->conninfo.connection);
4550 gst_rtsp_message_unset (&request);
4557 GST_WARNING_OBJECT (src, "no control url to send keepalive");
4562 gchar *str = gst_rtsp_strresult (res);
4564 gst_rtsp_message_unset (&request);
4565 GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
4566 ("Could not send keep-alive. (%s)", str));
4572 static GstFlowReturn
4573 gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
4575 GstFlowReturn ret = GST_FLOW_OK;
4577 GstRTSPStream *stream;
4578 GstPad *outpad = NULL;
4584 channel = message->type_data.data.channel;
4586 stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
4588 goto unknown_stream;
4590 if (channel == stream->channel[0]) {
4591 outpad = stream->channelpad[0];
4593 } else if (channel == stream->channel[1]) {
4594 outpad = stream->channelpad[1];
4600 /* take a look at the body to figure out what we have */
4601 gst_rtsp_message_get_body (message, &data, &size);
4603 goto invalid_length;
4605 /* channels are not correct on some servers, do extra check */
4606 if (data[1] >= 200 && data[1] <= 204) {
4607 /* hmm RTCP message switch to the RTCP pad of the same stream. */
4608 outpad = stream->channelpad[1];
4612 /* we have no clue what this is, just ignore then. */
4614 goto unknown_stream;
4616 /* take the message body for further processing */
4617 gst_rtsp_message_steal_body (message, &data, &size);
4619 /* strip the trailing \0 */
4622 buf = gst_buffer_new ();
4623 gst_buffer_append_memory (buf,
4624 gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
4626 /* don't need message anymore */
4627 gst_rtsp_message_unset (message);
4629 GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
4632 if (src->need_activate) {
4638 guint group_id = gst_util_group_id_next ();
4640 /* generate an SHA256 sum of the URI */
4641 cs = g_checksum_new (G_CHECKSUM_SHA256);
4642 uri = src->conninfo.location;
4643 g_checksum_update (cs, (const guchar *) uri, strlen (uri));
4645 for (streams = src->streams; streams; streams = g_list_next (streams)) {
4646 GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
4650 g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
4651 event = gst_event_new_stream_start (stream_id);
4652 gst_event_set_group_id (event, group_id);
4655 gst_rtspsrc_stream_push_event (src, ostream, event);
4657 if ((caps = stream_get_caps_for_pt (ostream, ostream->default_pt))) {
4658 /* only streams that have a connection to the outside world */
4659 if (ostream->setup) {
4660 if (ostream->udpsrc[0]) {
4661 gst_element_send_event (ostream->udpsrc[0],
4662 gst_event_new_caps (caps));
4663 } else if (ostream->channelpad[0]) {
4664 if (GST_PAD_IS_SRC (ostream->channelpad[0]))
4665 gst_pad_push_event (ostream->channelpad[0],
4666 gst_event_new_caps (caps));
4668 gst_pad_send_event (ostream->channelpad[0],
4669 gst_event_new_caps (caps));
4672 caps = gst_caps_new_empty_simple ("application/x-rtcp");
4674 if (ostream->udpsrc[1]) {
4675 gst_element_send_event (ostream->udpsrc[1],
4676 gst_event_new_caps (caps));
4677 } else if (ostream->channelpad[1]) {
4678 if (GST_PAD_IS_SRC (ostream->channelpad[1]))
4679 gst_pad_push_event (ostream->channelpad[1],
4680 gst_event_new_caps (caps));
4682 gst_pad_send_event (ostream->channelpad[1],
4683 gst_event_new_caps (caps));
4686 gst_caps_unref (caps);
4690 g_checksum_free (cs);
4692 gst_rtspsrc_activate_streams (src);
4693 src->need_activate = FALSE;
4694 src->need_segment = TRUE;
4697 if (src->base_time == -1) {
4698 /* Take current running_time. This timestamp will be put on
4699 * the first buffer of each stream because we are a live source and so we
4700 * timestamp with the running_time. When we are dealing with TCP, we also
4701 * only timestamp the first buffer (using the DISCONT flag) because a server
4702 * typically bursts data, for which we don't want to compensate by speeding
4703 * up the media. The other timestamps will be interpollated from this one
4704 * using the RTP timestamps. */
4705 GST_OBJECT_LOCK (src);
4706 if (GST_ELEMENT_CLOCK (src)) {
4708 GstClockTime base_time;
4710 now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
4711 base_time = GST_ELEMENT_CAST (src)->base_time;
4713 src->base_time = now - base_time;
4715 GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
4716 GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
4718 GST_OBJECT_UNLOCK (src);
4721 /* If needed send a new segment, don't forget we are live and buffer are
4722 * timestamped with running time */
4723 if (src->need_segment) {
4725 src->need_segment = FALSE;
4726 gst_segment_init (&segment, GST_FORMAT_TIME);
4727 gst_rtspsrc_push_event (src, gst_event_new_segment (&segment));
4730 if (stream->discont && !is_rtcp) {
4731 /* mark first RTP buffer as discont */
4732 GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
4733 stream->discont = FALSE;
4734 /* first buffer gets the timestamp, other buffers are not timestamped and
4735 * their presentation time will be interpollated from the rtp timestamps. */
4736 GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
4737 GST_TIME_ARGS (src->base_time));
4739 GST_BUFFER_TIMESTAMP (buf) = src->base_time;
4742 /* chain to the peer pad */
4743 if (GST_PAD_IS_SINK (outpad))
4744 ret = gst_pad_chain (outpad, buf);
4746 ret = gst_pad_push (outpad, buf);
4749 /* combine all stream flows for the data transport */
4750 ret = gst_rtspsrc_combine_flows (src, stream, ret);
4757 GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
4758 gst_rtsp_message_unset (message);
4763 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4764 ("Short message received, ignoring."));
4765 gst_rtsp_message_unset (message);
4770 static GstFlowReturn
4771 gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
4773 GstRTSPMessage message = { 0 };
4775 GstFlowReturn ret = GST_FLOW_OK;
4776 GTimeVal tv_timeout;
4779 /* get the next timeout interval */
4780 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4782 /* see if the timeout period expired */
4783 if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
4784 GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
4785 /* send keep-alive, only act on interrupt, a warning will be posted for
4787 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4789 /* get new timeout */
4790 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4793 GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
4794 tv_timeout.tv_sec, tv_timeout.tv_usec);
4796 /* protect the connection with the connection lock so that we can see when
4797 * we are finished doing server communication */
4799 gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4800 &message, src->ptcp_timeout);
4804 GST_DEBUG_OBJECT (src, "we received a server message");
4806 case GST_RTSP_EINTR:
4807 /* we got interrupted this means we need to stop */
4809 case GST_RTSP_ETIMEOUT:
4810 /* no reply, send keep alive */
4811 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4812 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4816 /* go EOS when the server closed the connection */
4822 switch (message.type) {
4823 case GST_RTSP_MESSAGE_REQUEST:
4824 /* server sends us a request message, handle it */
4826 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4828 if (res == GST_RTSP_EEOF)
4831 goto handle_request_failed;
4833 case GST_RTSP_MESSAGE_RESPONSE:
4834 /* we ignore response messages */
4835 GST_DEBUG_OBJECT (src, "ignoring response message");
4837 gst_rtsp_message_dump (&message);
4839 case GST_RTSP_MESSAGE_DATA:
4840 GST_DEBUG_OBJECT (src, "got data message");
4841 ret = gst_rtspsrc_handle_data (src, &message);
4842 if (ret != GST_FLOW_OK)
4843 goto handle_data_failed;
4846 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4851 g_assert_not_reached ();
4856 GST_DEBUG_OBJECT (src, "we got an eof from the server");
4857 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4858 ("The server closed the connection."));
4859 src->conninfo.connected = FALSE;
4860 gst_rtsp_message_unset (&message);
4861 return GST_FLOW_EOS;
4865 gst_rtsp_message_unset (&message);
4866 GST_DEBUG_OBJECT (src, "got interrupted");
4867 return GST_FLOW_FLUSHING;
4871 gchar *str = gst_rtsp_strresult (res);
4873 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
4874 ("Could not receive message. (%s)", str));
4877 gst_rtsp_message_unset (&message);
4878 return GST_FLOW_ERROR;
4880 handle_request_failed:
4882 gchar *str = gst_rtsp_strresult (res);
4884 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
4885 ("Could not handle server message. (%s)", str));
4887 gst_rtsp_message_unset (&message);
4888 return GST_FLOW_ERROR;
4892 GST_DEBUG_OBJECT (src, "could no handle data message");
4897 static GstFlowReturn
4898 gst_rtspsrc_loop_udp (GstRTSPSrc * src)
4901 GstRTSPMessage message = { 0 };
4905 GTimeVal tv_timeout;
4907 /* get the next timeout interval */
4908 gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
4910 GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
4911 (gint) tv_timeout.tv_sec);
4913 gst_rtsp_message_unset (&message);
4915 /* we should continue reading the TCP socket because the server might
4916 * send us requests. When the session timeout expires, we need to send a
4917 * keep-alive request to keep the session open. */
4918 res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
4919 &message, &tv_timeout);
4923 GST_DEBUG_OBJECT (src, "we received a server message");
4925 case GST_RTSP_EINTR:
4926 /* we got interrupted, see what we have to do */
4928 case GST_RTSP_ETIMEOUT:
4929 /* send keep-alive, ignore the result, a warning will be posted. */
4930 GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
4931 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4935 /* server closed the connection. not very fatal for UDP, reconnect and
4936 * see what happens. */
4937 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4938 ("The server closed the connection."));
4939 if (src->udp_reconnect) {
4941 gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
4948 GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
4950 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
4951 ("Unhandled return value %d.", res));
4955 switch (message.type) {
4956 case GST_RTSP_MESSAGE_REQUEST:
4957 /* server sends us a request message, handle it */
4959 gst_rtspsrc_handle_request (src, src->conninfo.connection,
4961 if (res == GST_RTSP_EEOF)
4964 goto handle_request_failed;
4966 case GST_RTSP_MESSAGE_RESPONSE:
4967 /* we ignore response and data messages */
4968 GST_DEBUG_OBJECT (src, "ignoring response message");
4970 gst_rtsp_message_dump (&message);
4971 if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
4972 GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
4973 if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
4974 GST_DEBUG_OBJECT (src, "so retrying keep-alive");
4975 if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
4982 case GST_RTSP_MESSAGE_DATA:
4983 /* we ignore response and data messages */
4984 GST_DEBUG_OBJECT (src, "ignoring data message");
4987 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
4992 g_assert_not_reached ();
4994 /* we get here when the connection got interrupted */
4997 gst_rtsp_message_unset (&message);
4998 GST_DEBUG_OBJECT (src, "got interrupted");
4999 return GST_FLOW_FLUSHING;
5003 gchar *str = gst_rtsp_strresult (res);
5006 src->conninfo.connected = FALSE;
5007 if (res != GST_RTSP_EINTR) {
5008 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
5009 ("Could not connect to server. (%s)", str));
5011 ret = GST_FLOW_ERROR;
5013 ret = GST_FLOW_FLUSHING;
5019 gchar *str = gst_rtsp_strresult (res);
5021 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5022 ("Could not receive message. (%s)", str));
5024 return GST_FLOW_ERROR;
5026 handle_request_failed:
5028 gchar *str = gst_rtsp_strresult (res);
5031 gst_rtsp_message_unset (&message);
5032 if (res != GST_RTSP_EINTR) {
5033 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5034 ("Could not handle server message. (%s)", str));
5036 ret = GST_FLOW_ERROR;
5038 ret = GST_FLOW_FLUSHING;
5044 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5045 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5046 ("The server closed the connection."));
5047 src->conninfo.connected = FALSE;
5048 gst_rtsp_message_unset (&message);
5049 return GST_FLOW_EOS;
5053 static GstRTSPResult
5054 gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
5056 GstRTSPResult res = GST_RTSP_OK;
5059 GST_DEBUG_OBJECT (src, "doing reconnect");
5061 GST_OBJECT_LOCK (src);
5062 /* only restart when the pads were not yet activated, else we were
5063 * streaming over UDP */
5064 restart = src->need_activate;
5065 GST_OBJECT_UNLOCK (src);
5067 /* no need to restart, we're done */
5071 /* we can try only TCP now */
5072 src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
5074 /* close and cleanup our state */
5075 if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
5078 /* see if we have TCP left to try. Also don't try TCP when we were configured
5080 if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
5083 /* We post a warning message now to inform the user
5084 * that nothing happened. It's most likely a firewall thing. */
5085 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5086 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5087 "firewall is blocking it. Retrying using a TCP connection.",
5088 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5090 /* open new connection using tcp */
5091 if (gst_rtspsrc_open (src, async) < 0)
5094 /* start playback */
5095 if (gst_rtspsrc_play (src, &src->segment, async) < 0)
5104 src->cur_protocols = 0;
5105 /* no transport possible, post an error and stop */
5106 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5107 ("Could not receive any UDP packets for %.4f seconds, maybe your "
5108 "firewall is blocking it. No other protocols to try.",
5109 gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
5110 return GST_RTSP_ERROR;
5114 GST_DEBUG_OBJECT (src, "open failed");
5119 GST_DEBUG_OBJECT (src, "play failed");
5125 gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
5129 GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
5132 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
5135 GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
5138 GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
5146 gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
5150 GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
5153 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
5156 GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
5159 GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
5167 gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
5171 GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
5174 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
5177 GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
5180 GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
5188 gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
5192 GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
5195 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
5198 GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
5201 GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
5209 gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
5211 if (ret == GST_RTSP_OK)
5212 gst_rtspsrc_loop_complete_cmd (src, cmd);
5213 else if (ret == GST_RTSP_EINTR)
5214 gst_rtspsrc_loop_cancel_cmd (src, cmd);
5216 gst_rtspsrc_loop_error_cmd (src, cmd);
5220 gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
5223 gboolean flushed = FALSE;
5225 /* start new request */
5226 gst_rtspsrc_loop_start_cmd (src, cmd);
5228 GST_DEBUG_OBJECT (src, "sending cmd %s", cmd_to_string (cmd));
5230 GST_OBJECT_LOCK (src);
5231 old = src->pending_cmd;
5232 if (old == CMD_RECONNECT) {
5233 GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
5234 cmd = CMD_RECONNECT;
5236 if (old != CMD_WAIT) {
5237 src->pending_cmd = CMD_WAIT;
5238 GST_OBJECT_UNLOCK (src);
5239 /* cancel previous request */
5240 GST_DEBUG_OBJECT (src, "cancel previous request %s", cmd_to_string (old));
5241 gst_rtspsrc_loop_cancel_cmd (src, old);
5242 GST_OBJECT_LOCK (src);
5244 src->pending_cmd = cmd;
5245 /* interrupt if allowed */
5246 if (src->busy_cmd & mask) {
5247 GST_DEBUG_OBJECT (src, "connection flush busy %s",
5248 cmd_to_string (src->busy_cmd));
5249 gst_rtspsrc_connection_flush (src, TRUE);
5252 GST_DEBUG_OBJECT (src, "not interrupting busy cmd %s",
5253 cmd_to_string (src->busy_cmd));
5256 gst_task_start (src->task);
5257 GST_OBJECT_UNLOCK (src);
5263 gst_rtspsrc_loop (GstRTSPSrc * src)
5267 if (!src->conninfo.connection || !src->conninfo.connected)
5270 if (src->interleaved)
5271 ret = gst_rtspsrc_loop_interleaved (src);
5273 ret = gst_rtspsrc_loop_udp (src);
5275 if (ret != GST_FLOW_OK)
5283 GST_WARNING_OBJECT (src, "we are not connected");
5284 ret = GST_FLOW_FLUSHING;
5289 const gchar *reason = gst_flow_get_name (ret);
5291 GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
5292 src->running = FALSE;
5293 if (ret == GST_FLOW_EOS) {
5294 /* perform EOS logic */
5295 if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
5296 gst_element_post_message (GST_ELEMENT_CAST (src),
5297 gst_message_new_segment_done (GST_OBJECT_CAST (src),
5298 src->segment.format, src->segment.position));
5299 gst_rtspsrc_push_event (src,
5300 gst_event_new_segment_done (src->segment.format,
5301 src->segment.position));
5303 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5305 } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
5306 /* for fatal errors we post an error message, post the error before the
5307 * EOS so the app knows about the error first. */
5308 GST_ELEMENT_ERROR (src, STREAM, FAILED,
5309 ("Internal data flow error."),
5310 ("streaming task paused, reason %s (%d)", reason, ret));
5311 gst_rtspsrc_push_event (src, gst_event_new_eos ());
5313 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
5318 #ifndef GST_DISABLE_GST_DEBUG
5319 static const gchar *
5320 gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
5324 while (method != 0) {
5341 static const gchar *
5342 gst_rtspsrc_skip_lws (const gchar * s)
5344 while (g_ascii_isspace (*s))
5349 static const gchar *
5350 gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
5352 while (s > start && g_ascii_isspace (*(s - 1)))
5357 static const gchar *
5358 gst_rtspsrc_skip_commas (const gchar * s)
5360 /* The grammar allows for multiple commas */
5361 while (g_ascii_isspace (*s) || *s == ',')
5366 static const gchar *
5367 gst_rtspsrc_skip_item (const gchar * s)
5369 gboolean quoted = FALSE;
5370 const gchar *start = s;
5372 /* A list item ends at the last non-whitespace character
5373 * before a comma which is not inside a quoted-string. Or at
5374 * the end of the string.
5380 if (*s == '\\' && *(s + 1))
5389 return gst_rtspsrc_unskip_lws (s, start);
5393 gst_rtsp_decode_quoted_string (gchar * quoted_string)
5397 src = quoted_string + 1;
5398 dst = quoted_string;
5399 while (*src && *src != '"') {
5400 if (*src == '\\' && *(src + 1))
5407 /* Extract the authentication tokens that the server provided for each method
5408 * into an array of structures and give those to the connection object.
5411 gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
5412 const gchar * header, gboolean * stale)
5414 GSList *list = NULL, *iter;
5416 gchar *item, *eq, *name_end, *value;
5418 g_return_if_fail (stale != NULL);
5420 gst_rtsp_connection_clear_auth_params (conn);
5423 /* Parse a header whose content is described by RFC2616 as
5424 * "#something", where "something" does not itself contain commas,
5425 * except as part of quoted-strings, into a list of allocated strings.
5427 header = gst_rtspsrc_skip_commas (header);
5429 end = gst_rtspsrc_skip_item (header);
5430 list = g_slist_prepend (list, g_strndup (header, end - header));
5431 header = gst_rtspsrc_skip_commas (end);
5436 list = g_slist_reverse (list);
5437 for (iter = list; iter; iter = iter->next) {
5440 eq = strchr (item, '=');
5442 name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
5443 if (name_end == item) {
5444 /* That's no good... */
5451 value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
5453 gst_rtsp_decode_quoted_string (value);
5457 if (value && strcmp (item, "stale") == 0 && strcmp (value, "TRUE") == 0)
5459 gst_rtsp_connection_set_auth_param (conn, item, value);
5463 g_slist_free (list);
5466 /* Parse a WWW-Authenticate Response header and determine the
5467 * available authentication methods
5469 * This code should also cope with the fact that each WWW-Authenticate
5470 * header can contain multiple challenge methods + tokens
5472 * At the moment, for Basic auth, we just do a minimal check and don't
5473 * even parse out the realm */
5475 gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
5476 GstRTSPConnection * conn, gboolean * stale)
5480 g_return_if_fail (hdr != NULL);
5481 g_return_if_fail (methods != NULL);
5482 g_return_if_fail (stale != NULL);
5484 /* Skip whitespace at the start of the string */
5485 for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
5487 if (g_ascii_strncasecmp (start, "basic", 5) == 0)
5488 *methods |= GST_RTSP_AUTH_BASIC;
5489 else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
5490 *methods |= GST_RTSP_AUTH_DIGEST;
5491 gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
5496 * gst_rtspsrc_setup_auth:
5497 * @src: the rtsp source
5499 * Configure a username and password and auth method on the
5500 * connection object based on a response we received from the
5503 * Currently, this requires that a username and password were supplied
5504 * in the uri. In the future, they may be requested on demand by sending
5505 * a message up the bus.
5507 * Returns: TRUE if authentication information could be set up correctly.
5510 gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
5514 GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
5515 GstRTSPAuthMethod method;
5516 GstRTSPResult auth_result;
5518 GstRTSPConnection *conn;
5520 gboolean stale = FALSE;
5522 conn = src->conninfo.connection;
5524 /* Identify the available auth methods and see if any are supported */
5525 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
5526 &hdr, 0) == GST_RTSP_OK) {
5527 gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
5530 if (avail_methods == GST_RTSP_AUTH_NONE)
5531 goto no_auth_available;
5533 /* For digest auth, if the response indicates that the session
5534 * data are stale, we just update them in the connection object and
5535 * return TRUE to retry the request */
5537 src->tried_url_auth = FALSE;
5539 url = gst_rtsp_connection_get_url (conn);
5541 /* Do we have username and password available? */
5542 if (url != NULL && !src->tried_url_auth && url->user != NULL
5543 && url->passwd != NULL) {
5546 src->tried_url_auth = TRUE;
5547 GST_DEBUG_OBJECT (src,
5548 "Attempting authentication using credentials from the URL");
5550 user = src->user_id;
5551 pass = src->user_pw;
5552 GST_DEBUG_OBJECT (src,
5553 "Attempting authentication using credentials from the properties");
5556 /* FIXME: If the url didn't contain username and password or we tried them
5557 * already, request a username and passwd from the application via some kind
5558 * of credentials request message */
5560 /* If we don't have a username and passwd at this point, bail out. */
5561 if (user == NULL || pass == NULL)
5564 /* Try to configure for each available authentication method, strongest to
5566 for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
5567 /* Check if this method is available on the server */
5568 if ((method & avail_methods) == 0)
5571 /* Pass the credentials to the connection to try on the next request */
5572 auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
5573 /* INVAL indicates an invalid username/passwd were supplied, so we'll just
5574 * ignore it and end up retrying later */
5575 if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
5576 GST_DEBUG_OBJECT (src, "Attempting %s authentication",
5577 gst_rtsp_auth_method_to_string (method));
5582 if (method == GST_RTSP_AUTH_NONE)
5583 goto no_auth_available;
5589 /* Output an error indicating that we couldn't connect because there were
5590 * no supported authentication protocols */
5591 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5592 ("No supported authentication protocol was found"));
5597 /* We don't fire an error message, we just return FALSE and let the
5598 * normal NOT_AUTHORIZED error be propagated */
5603 static GstRTSPResult
5604 gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5605 GstRTSPMessage * request, GstRTSPMessage * response,
5606 GstRTSPStatusCode * code)
5609 GstRTSPStatusCode thecode;
5610 gchar *content_base = NULL;
5614 if (!src->short_header)
5615 gst_rtsp_ext_list_before_send (src->extensions, request);
5617 GST_DEBUG_OBJECT (src, "sending message");
5620 gst_rtsp_message_dump (request);
5622 res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
5626 gst_rtsp_connection_reset_timeout (conn);
5629 res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
5634 gst_rtsp_message_dump (response);
5636 switch (response->type) {
5637 case GST_RTSP_MESSAGE_REQUEST:
5638 res = gst_rtspsrc_handle_request (src, conn, response);
5639 if (res == GST_RTSP_EEOF)
5642 goto handle_request_failed;
5644 case GST_RTSP_MESSAGE_RESPONSE:
5645 /* ok, a response is good */
5646 GST_DEBUG_OBJECT (src, "received response message");
5648 case GST_RTSP_MESSAGE_DATA:
5649 /* get next response */
5650 GST_DEBUG_OBJECT (src, "handle data response message");
5651 gst_rtspsrc_handle_data (src, response);
5654 GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
5659 thecode = response->type_data.response.code;
5661 GST_DEBUG_OBJECT (src, "got response message %d", thecode);
5663 /* if the caller wanted the result code, we store it. */
5667 /* If the request didn't succeed, bail out before doing any more */
5668 if (thecode != GST_RTSP_STS_OK)
5671 /* store new content base if any */
5672 gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
5675 g_free (src->content_base);
5676 src->content_base = g_strdup (content_base);
5678 gst_rtsp_ext_list_after_send (src->extensions, request, response);
5685 gchar *str = gst_rtsp_strresult (res);
5687 if (res != GST_RTSP_EINTR) {
5688 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
5689 ("Could not send message. (%s)", str));
5691 GST_WARNING_OBJECT (src, "send interrupted");
5700 GST_WARNING_OBJECT (src, "server closed connection");
5701 if ((try == 0) && !src->interleaved && src->udp_reconnect) {
5703 /* if reconnect succeeds, try again */
5705 gst_rtsp_conninfo_reconnect (src, &src->conninfo,
5709 /* only try once after reconnect, then fallthrough and error out */
5712 gchar *str = gst_rtsp_strresult (res);
5714 if (res != GST_RTSP_EINTR) {
5715 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5716 ("Could not receive message. (%s)", str));
5718 GST_WARNING_OBJECT (src, "receive interrupted");
5726 handle_request_failed:
5728 /* ERROR was posted */
5729 gst_rtsp_message_unset (response);
5734 GST_DEBUG_OBJECT (src, "we got an eof from the server");
5735 GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
5736 ("The server closed the connection."));
5737 gst_rtsp_message_unset (response);
5744 * @src: the rtsp source
5745 * @conn: the connection to send on
5746 * @request: must point to a valid request
5747 * @response: must point to an empty #GstRTSPMessage
5748 * @code: an optional code result
5750 * send @request and retrieve the response in @response. optionally @code can be
5751 * non-NULL in which case it will contain the status code of the response.
5753 * If This function returns #GST_RTSP_OK, @response will contain a valid response
5754 * message that should be cleaned with gst_rtsp_message_unset() after usage.
5756 * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
5757 * @response message) if the response code was not 200 (OK).
5759 * If the attempt results in an authentication failure, then this will attempt
5760 * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
5763 * Returns: #GST_RTSP_OK if the processing was successful.
5765 static GstRTSPResult
5766 gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
5767 GstRTSPMessage * request, GstRTSPMessage * response,
5768 GstRTSPStatusCode * code)
5770 GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
5771 GstRTSPResult res = GST_RTSP_ERROR;
5774 GstRTSPMethod method = GST_RTSP_INVALID;
5780 /* make sure we don't loop forever */
5784 /* save method so we can disable it when the server complains */
5785 method = request->type_data.request.method;
5788 gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
5792 case GST_RTSP_STS_UNAUTHORIZED:
5793 if (gst_rtspsrc_setup_auth (src, response)) {
5794 /* Try the request/response again after configuring the auth info
5802 } while (retry == TRUE);
5804 /* If the user requested the code, let them handle errors, otherwise
5805 * post an error below */
5808 else if (int_code != GST_RTSP_STS_OK)
5809 goto error_response;
5816 GST_DEBUG_OBJECT (src, "got error %d", res);
5821 res = GST_RTSP_ERROR;
5823 switch (response->type_data.response.code) {
5824 case GST_RTSP_STS_NOT_FOUND:
5825 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
5826 response->type_data.response.reason));
5828 case GST_RTSP_STS_UNAUTHORIZED:
5829 GST_ELEMENT_ERROR (src, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
5830 response->type_data.response.reason));
5832 case GST_RTSP_STS_MOVED_PERMANENTLY:
5833 case GST_RTSP_STS_MOVE_TEMPORARILY:
5835 gchar *new_location;
5836 GstRTSPLowerTrans transports;
5838 GST_DEBUG_OBJECT (src, "got redirection");
5839 /* if we don't have a Location Header, we must error */
5840 if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
5841 &new_location, 0) < 0)
5844 /* When we receive a redirect result, we go back to the INIT state after
5845 * parsing the new URI. The caller should do the needed steps to issue
5846 * a new setup when it detects this state change. */
5847 GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
5849 /* save current transports */
5850 if (src->conninfo.url)
5851 transports = src->conninfo.url->transports;
5853 transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
5855 gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
5857 /* set old transports */
5858 if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
5859 src->conninfo.url->transports = transports;
5861 src->need_redirect = TRUE;
5862 src->state = GST_RTSP_STATE_INIT;
5866 case GST_RTSP_STS_NOT_ACCEPTABLE:
5867 case GST_RTSP_STS_NOT_IMPLEMENTED:
5868 case GST_RTSP_STS_METHOD_NOT_ALLOWED:
5869 GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
5870 gst_rtsp_method_as_text (method));
5871 src->methods &= ~method;
5875 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
5876 ("Got error response: %d (%s).", response->type_data.response.code,
5877 response->type_data.response.reason));
5880 /* if we return ERROR we should unset the response ourselves */
5881 if (res == GST_RTSP_ERROR)
5882 gst_rtsp_message_unset (response);
5888 static GstRTSPResult
5889 gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
5890 GstRTSPMessage * response, GstRTSPSrc * src)
5892 return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
5897 /* parse the response and collect all the supported methods. We need this
5898 * information so that we don't try to send an unsupported request to the
5902 gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
5904 GstRTSPHeaderField field;
5908 /* reset supported methods */
5911 /* Try Allow Header first */
5912 field = GST_RTSP_HDR_ALLOW;
5915 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5916 if (indx == 0 && !respoptions) {
5917 /* if no Allow header was found then try the Public header... */
5918 field = GST_RTSP_HDR_PUBLIC;
5919 gst_rtsp_message_get_header (response, field, &respoptions, indx);
5924 src->methods |= gst_rtsp_options_from_text (respoptions);
5929 if (src->methods == 0) {
5930 /* neither Allow nor Public are required, assume the server supports
5931 * at least DESCRIBE, SETUP, we always assume it supports PLAY as
5933 GST_DEBUG_OBJECT (src, "could not get OPTIONS");
5934 src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
5936 /* always assume PLAY, FIXME, extensions should be able to override
5938 src->methods |= GST_RTSP_PLAY;
5939 /* also assume it will support Range */
5940 src->seekable = TRUE;
5942 /* we need describe and setup */
5943 if (!(src->methods & GST_RTSP_DESCRIBE))
5945 if (!(src->methods & GST_RTSP_SETUP))
5953 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5954 ("Server does not support DESCRIBE."));
5959 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
5960 ("Server does not support SETUP."));
5965 /* masks to be kept in sync with the hardcoded protocol order of preference
5967 static const guint protocol_masks[] = {
5968 GST_RTSP_LOWER_TRANS_UDP,
5969 GST_RTSP_LOWER_TRANS_UDP_MCAST,
5970 GST_RTSP_LOWER_TRANS_TCP,
5974 static GstRTSPResult
5975 gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
5976 GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
5980 gboolean add_udp_str;
5985 gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
5990 GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
5992 /* extension listed transports, use those */
5993 if (*transports != NULL)
5996 /* it's the default */
5997 add_udp_str = FALSE;
5999 /* the default RTSP transports */
6000 result = g_string_new ("RTP");
6003 case GST_RTSP_PROFILE_AVP:
6004 g_string_append (result, "/AVP");
6006 case GST_RTSP_PROFILE_SAVP:
6007 g_string_append (result, "/SAVP");
6009 case GST_RTSP_PROFILE_AVPF:
6010 g_string_append (result, "/AVPF");
6012 case GST_RTSP_PROFILE_SAVPF:
6013 g_string_append (result, "/SAVPF");
6019 if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
6020 GST_DEBUG_OBJECT (src, "adding UDP unicast");
6022 g_string_append (result, "/UDP");
6023 g_string_append (result, ";unicast;client_port=%%u1-%%u2");
6024 } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
6025 GST_DEBUG_OBJECT (src, "adding UDP multicast");
6026 /* we don't have to allocate any UDP ports yet, if the selected transport
6027 * turns out to be multicast we can create them and join the multicast
6028 * group indicated in the transport reply */
6030 g_string_append (result, "/UDP");
6031 g_string_append (result, ";multicast");
6032 if (src->next_port_num != 0) {
6033 if (src->client_port_range.max > 0 &&
6034 src->next_port_num >= src->client_port_range.max)
6037 g_string_append_printf (result, ";client_port=%d-%d",
6038 src->next_port_num, src->next_port_num + 1);
6040 } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
6041 GST_DEBUG_OBJECT (src, "adding TCP");
6043 g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
6045 *transports = g_string_free (result, FALSE);
6047 GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
6054 GST_ERROR ("extension gave error %d", res);
6059 GST_ERROR ("no more ports available");
6060 return GST_RTSP_ERROR;
6064 static GstRTSPResult
6065 gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
6066 gint orig_rtpport, gint orig_rtcpport)
6069 gint nr_udp, nr_int;
6071 gint rtpport = 0, rtcpport = 0;
6074 src = stream->parent;
6076 /* find number of placeholders first */
6077 if (strstr (*transports, "%%i2"))
6079 else if (strstr (*transports, "%%i1"))
6084 if (strstr (*transports, "%%u2"))
6086 else if (strstr (*transports, "%%u1"))
6091 if (nr_udp == 0 && nr_int == 0)
6095 if (!orig_rtpport || !orig_rtcpport) {
6096 if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
6099 rtpport = orig_rtpport;
6100 rtcpport = orig_rtcpport;
6104 str = g_string_new ("");
6106 while ((next = strstr (p, "%%"))) {
6107 g_string_append_len (str, p, next - p);
6108 if (next[2] == 'u') {
6110 g_string_append_printf (str, "%d", rtpport);
6111 else if (next[3] == '2')
6112 g_string_append_printf (str, "%d", rtcpport);
6114 if (next[2] == 'i') {
6116 g_string_append_printf (str, "%d", src->free_channel);
6117 else if (next[3] == '2')
6118 g_string_append_printf (str, "%d", src->free_channel + 1);
6123 /* append final part */
6124 g_string_append (str, p);
6126 g_free (*transports);
6127 *transports = g_string_free (str, FALSE);
6135 GST_ERROR ("failed to allocate udp ports");
6136 return GST_RTSP_ERROR;
6141 enc_key_length_from_cipher_name (const gchar * cipher)
6143 if (g_strcmp0 (cipher, "aes-128-icm") == 0)
6144 return AES_128_KEY_LEN;
6145 else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
6146 return AES_256_KEY_LEN;
6148 GST_ERROR ("encryption algorithm '%s' not supported", cipher);
6154 auth_key_length_from_auth_name (const gchar * auth)
6156 if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
6157 return HMAC_32_KEY_LEN;
6158 else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
6159 return HMAC_80_KEY_LEN;
6161 GST_ERROR ("authentication algorithm '%s' not supported", auth);
6167 signal_get_srtcp_params (GstRTSPSrc * src, GstRTSPStream * stream)
6169 GstCaps *caps = NULL;
6171 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
6175 GST_DEBUG_OBJECT (src, "SRTP parameters received");
6181 default_srtcp_params (void)
6189 /* create a random key */
6190 key_data = g_malloc (KEY_SIZE);
6191 for (i = 0; i < KEY_SIZE; i += 4)
6192 GST_WRITE_UINT32_BE (key_data + i, g_random_int ());
6194 buf = gst_buffer_new_wrapped (key_data, KEY_SIZE);
6196 caps = gst_caps_new_simple ("application/x-srtp",
6197 "srtp-key", GST_TYPE_BUFFER, buf,
6198 "srtcp-cipher", G_TYPE_STRING, "aes-128-icm",
6199 "srtcp-auth", G_TYPE_STRING, "hmac-sha1-80", NULL);
6201 gst_buffer_unref (buf);
6207 gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
6210 gchar *result, *base64;
6213 GstMIKEYMessage *msg;
6214 GstMIKEYPayload *payload, *pkd;
6220 const gchar *srtcpcipher, *srtcpauth;
6222 stream->srtcpparams = signal_get_srtcp_params (src, stream);
6223 if (stream->srtcpparams == NULL)
6224 stream->srtcpparams = default_srtcp_params ();
6226 s = gst_caps_get_structure (stream->srtcpparams, 0);
6228 srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
6229 srtcpauth = gst_structure_get_string (s, "srtcp-auth");
6230 val = gst_structure_get_value (s, "srtp-key");
6232 if (srtcpcipher == NULL || srtcpauth == NULL || val == NULL) {
6233 GST_ERROR_OBJECT (src, "could not find the right SRTP parameters in caps");
6237 srtpkey = gst_value_get_buffer (val);
6239 msg = gst_mikey_message_new ();
6240 /* unencrypted MIKEY message, we send this over TLS so this is allowed */
6241 gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
6242 FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
6243 /* add policy '0' for our SSRC */
6244 gst_mikey_message_add_cs_srtp (msg, 0, stream->send_ssrc, 0);
6245 /* timestamp is now */
6246 gst_mikey_message_add_t_now_ntp_utc (msg);
6247 /* add some random data */
6248 gst_mikey_message_add_rand_len (msg, 16);
6250 /* the policy '0' is SRTP */
6251 payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
6252 gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
6254 /* only AES-CM is supported */
6256 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
6257 /* encryption key length */
6258 byte = enc_key_length_from_cipher_name (srtcpcipher);
6259 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
6261 /* only HMAC-SHA1 */
6262 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
6264 /* authentication key length */
6265 byte = auth_key_length_from_auth_name (srtcpauth);
6266 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
6268 /* we enable encryption on RTP and RTCP */
6269 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
6271 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
6273 /* we enable authentication on RTP and RTCP */
6274 gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
6276 gst_mikey_message_add_payload (msg, payload);
6278 /* make unencrypted KEMAC */
6279 payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
6280 gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL);
6281 /* add the key in KEMAC */
6282 pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
6283 gst_buffer_map (srtpkey, &info, GST_MAP_READ);
6284 gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
6286 gst_buffer_unmap (srtpkey, &info);
6287 gst_mikey_payload_kemac_add_sub (payload, pkd);
6288 gst_mikey_message_add_payload (msg, payload);
6290 /* now serialize this to bytes */
6291 bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
6292 gst_mikey_message_unref (msg);
6293 /* and make it into base64 */
6294 data = g_bytes_get_data (bytes, &size);
6295 base64 = g_base64_encode (data, size);
6296 g_bytes_unref (bytes);
6298 result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
6299 stream->conninfo.location, base64);
6306 /* Perform the SETUP request for all the streams.
6308 * We ask the server for a specific transport, which initially includes all the
6309 * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
6310 * two local UDP ports that we send to the server.
6312 * Once the server replied with a transport, we configure the other streams
6313 * with the same transport.
6315 * This function will also configure the stream for the selected transport,
6316 * which basically means creating the pipeline.
6318 static GstRTSPResult
6319 gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
6322 GstRTSPResult res = GST_RTSP_ERROR;
6323 GstRTSPMessage request = { 0 };
6324 GstRTSPMessage response = { 0 };
6325 GstRTSPStream *stream = NULL;
6326 GstRTSPLowerTrans protocols;
6327 GstRTSPStatusCode code;
6328 gboolean unsupported_real = FALSE;
6329 gint rtpport, rtcpport;
6333 if (src->conninfo.connection) {
6334 url = gst_rtsp_connection_get_url (src->conninfo.connection);
6335 /* we initially allow all configured lower transports. based on the URL
6336 * transports and the replies from the server we narrow them down. */
6337 protocols = url->transports & src->cur_protocols;
6340 protocols = src->cur_protocols;
6346 /* reset some state */
6347 src->free_channel = 0;
6348 src->interleaved = FALSE;
6349 src->need_activate = FALSE;
6350 /* keep track of next port number, 0 is random */
6351 src->next_port_num = src->client_port_range.min;
6352 rtpport = rtcpport = 0;
6354 if (G_UNLIKELY (src->streams == NULL))
6357 for (walk = src->streams; walk; walk = g_list_next (walk)) {
6358 GstRTSPConnection *conn;
6365 stream = (GstRTSPStream *) walk->data;
6367 caps = stream_get_caps_for_pt (stream, stream->default_pt);
6369 GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
6373 if (stream->skipped) {
6374 GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
6378 /* see if we need to configure this stream */
6379 if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
6380 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
6385 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
6386 stream->id, caps, &selected);
6388 GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
6392 /* merge/overwrite global caps */
6397 s = gst_caps_get_structure (caps, 0);
6399 num = gst_structure_n_fields (src->props);
6400 for (j = 0; j < num; j++) {
6404 name = gst_structure_nth_field_name (src->props, j);
6405 val = gst_structure_get_value (src->props, name);
6406 gst_structure_set_value (s, name, val);
6408 GST_DEBUG_OBJECT (src, "copied %s", name);
6412 /* skip setup if we have no URL for it */
6413 if (stream->conninfo.location == NULL) {
6414 GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
6418 if (src->conninfo.connection == NULL) {
6419 if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
6420 GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
6423 conn = stream->conninfo.connection;
6425 conn = src->conninfo.connection;
6427 GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
6428 stream->conninfo.location);
6430 /* if we have a multicast connection, only suggest multicast from now on */
6431 if (stream->is_multicast)
6432 protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
6435 /* first selectable protocol */
6436 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6438 if (!protocol_masks[mask])
6442 GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
6443 protocol_masks[mask]);
6444 /* create a string with first transport in line */
6446 res = gst_rtspsrc_create_transports_string (src,
6447 protocols & protocol_masks[mask], stream->profile, &transports);
6448 if (res < 0 || transports == NULL)
6449 goto setup_transport_failed;
6451 if (strlen (transports) == 0) {
6452 g_free (transports);
6453 GST_DEBUG_OBJECT (src, "no transports found");
6458 GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
6460 /* replace placeholders with real values, this function will optionally
6461 * allocate UDP ports and other info needed to execute the setup request */
6462 res = gst_rtspsrc_prepare_transports (stream, &transports,
6463 retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
6465 g_free (transports);
6466 goto setup_transport_failed;
6469 GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
6471 /* create SETUP request */
6473 gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
6474 stream->conninfo.location);
6476 g_free (transports);
6477 goto create_request_failed;
6480 /* select transport */
6481 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
6484 if (stream->profile == GST_RTSP_PROFILE_SAVP ||
6485 stream->profile == GST_RTSP_PROFILE_SAVPF) {
6486 hval = gst_rtspsrc_stream_make_keymgmt (src, stream);
6487 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
6490 /* if the user wants a non default RTP packet size we add the blocksize
6492 if (src->rtp_blocksize > 0) {
6493 hval = g_strdup_printf ("%d", src->rtp_blocksize);
6494 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
6498 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
6501 /* handle the code ourselves */
6502 res = gst_rtspsrc_send (src, conn, &request, &response, &code);
6507 case GST_RTSP_STS_OK:
6509 case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
6510 gst_rtsp_message_unset (&request);
6511 gst_rtsp_message_unset (&response);
6512 /* cleanup of leftover transport */
6513 gst_rtspsrc_stream_free_udp (stream);
6514 /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
6515 * we might be in this case */
6516 if (stream->container && rtpport && rtcpport && !retry) {
6517 GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
6522 /* this transport did not go down well, but we may have others to try
6523 * that we did not send yet, try those and only give up then
6524 * but not without checking for lost cause/extension so we can
6525 * post a nicer/more useful error message later */
6526 if (!unsupported_real)
6527 unsupported_real = stream->is_real;
6528 /* select next available protocol, give up on this stream if none */
6530 while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
6532 if (!protocol_masks[mask] || unsupported_real)
6537 /* cleanup of leftover transport and move to the next stream */
6538 gst_rtspsrc_stream_free_udp (stream);
6539 goto response_error;
6542 /* parse response transport */
6544 gchar *resptrans = NULL;
6545 GstRTSPTransport transport = { 0 };
6547 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
6550 gst_rtspsrc_stream_free_udp (stream);
6554 /* parse transport, go to next stream on parse error */
6555 if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
6556 GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
6560 /* update allowed transports for other streams. once the transport of
6561 * one stream has been determined, we make sure that all other streams
6562 * are configured in the same way */
6563 switch (transport.lower_transport) {
6564 case GST_RTSP_LOWER_TRANS_TCP:
6565 GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
6566 protocols = GST_RTSP_LOWER_TRANS_TCP;
6567 src->interleaved = TRUE;
6568 /* update free channels */
6570 MAX (transport.interleaved.min, src->free_channel);
6572 MAX (transport.interleaved.max, src->free_channel);
6573 src->free_channel++;
6575 case GST_RTSP_LOWER_TRANS_UDP_MCAST:
6576 /* only allow multicast for other streams */
6577 GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
6578 protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
6579 /* if the server selected our ports, increment our counters so that
6580 * we select a new port later */
6581 if (src->next_port_num == transport.port.min &&
6582 src->next_port_num + 1 == transport.port.max) {
6583 src->next_port_num += 2;
6586 case GST_RTSP_LOWER_TRANS_UDP:
6587 /* only allow unicast for other streams */
6588 GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
6589 protocols = GST_RTSP_LOWER_TRANS_UDP;
6592 GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
6593 transport.lower_transport);
6597 if (!src->interleaved || !retry) {
6598 /* now configure the stream with the selected transport */
6599 if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
6600 GST_DEBUG_OBJECT (src,
6601 "could not configure stream %p transport, skipping stream",
6604 } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
6605 /* retain the first allocated UDP port pair */
6606 g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
6607 g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
6610 /* we need to activate at least one streams when we detect activity */
6611 src->need_activate = TRUE;
6613 /* stream is setup now */
6614 stream->setup = TRUE;
6619 GstRTSPStream *sskip;
6621 skip = g_list_next (skip);
6625 sskip = (GstRTSPStream *) skip->data;
6627 /* skip all streams with the same control url */
6628 if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
6629 GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
6630 sskip, sskip->conninfo.location);
6631 sskip->skipped = TRUE;
6636 /* clean up our transport struct */
6637 gst_rtsp_transport_init (&transport);
6638 /* clean up used RTSP messages */
6639 gst_rtsp_message_unset (&request);
6640 gst_rtsp_message_unset (&response);
6644 /* store the transport protocol that was configured */
6645 src->cur_protocols = protocols;
6647 gst_rtsp_ext_list_stream_select (src->extensions, url);
6649 /* if there is nothing to activate, error out */
6650 if (!src->need_activate)
6651 goto nothing_to_activate;
6658 /* no transport possible, post an error and stop */
6659 GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
6660 ("Could not connect to server, no protocols left"));
6661 return GST_RTSP_ERROR;
6665 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6666 ("SDP contains no streams"));
6667 return GST_RTSP_ERROR;
6669 create_request_failed:
6671 gchar *str = gst_rtsp_strresult (res);
6673 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
6674 ("Could not create request. (%s)", str));
6678 setup_transport_failed:
6680 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6681 ("Could not setup transport."));
6682 res = GST_RTSP_ERROR;
6687 const gchar *str = gst_rtsp_status_as_text (code);
6689 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6690 ("Error (%d): %s", code, GST_STR_NULL (str)));
6691 res = GST_RTSP_ERROR;
6696 gchar *str = gst_rtsp_strresult (res);
6698 if (res != GST_RTSP_EINTR) {
6699 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
6700 ("Could not send message. (%s)", str));
6702 GST_WARNING_OBJECT (src, "send interrupted");
6709 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
6710 ("Server did not select transport."));
6711 res = GST_RTSP_ERROR;
6714 nothing_to_activate:
6716 /* none of the available error codes is really right .. */
6717 if (unsupported_real) {
6718 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6719 (_("No supported stream was found. You might need to install a "
6720 "GStreamer RTSP extension plugin for Real media streams.")),
6723 GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
6724 (_("No supported stream was found. You might need to allow "
6725 "more transport protocols or may otherwise be missing "
6726 "the right GStreamer RTSP extension plugin.")), (NULL));
6728 return GST_RTSP_ERROR;
6732 gst_rtsp_message_unset (&request);
6733 gst_rtsp_message_unset (&response);
6739 gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
6740 GstSegment * segment)
6743 GstRTSPTimeRange *therange;
6746 gst_rtsp_range_free (src->range);
6748 if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
6749 GST_DEBUG_OBJECT (src, "parsed range %s", range);
6750 src->range = therange;
6752 GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
6754 gst_segment_init (segment, GST_FORMAT_TIME);
6758 GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d, max %f ",
6759 therange->min.type, therange->min.seconds, therange->max.type,
6760 therange->max.seconds);
6762 if (therange->min.type == GST_RTSP_TIME_NOW)
6764 else if (therange->min.type == GST_RTSP_TIME_END)
6767 seconds = therange->min.seconds * GST_SECOND;
6769 GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
6770 GST_TIME_ARGS (seconds));
6772 /* we need to start playback without clipping from the position reported by
6774 segment->start = seconds;
6775 segment->position = seconds;
6777 if (therange->max.type == GST_RTSP_TIME_NOW)
6779 else if (therange->max.type == GST_RTSP_TIME_END)
6782 seconds = therange->max.seconds * GST_SECOND;
6784 GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
6785 GST_TIME_ARGS (seconds));
6787 /* live (WMS) server might send overflowed large max as its idea of infinity,
6788 * compensate to prevent problems later on */
6789 if (seconds != -1 && seconds < 0) {
6791 GST_DEBUG_OBJECT (src, "insane range, set to NONE");
6794 /* live (WMS) might send min == max, which is not worth recording */
6795 if (segment->duration == -1 && seconds == segment->start)
6798 /* don't change duration with unknown value, we might have a valid value
6799 * there that we want to keep. */
6801 segment->duration = seconds;
6806 /* Parse clock profived by the server with following syntax:
6808 * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
6811 gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
6813 gboolean res = FALSE;
6815 if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
6816 gchar **fields = NULL, **parts = NULL;
6817 gchar *remote_ip, *str;
6819 GstClockTime base_time;
6822 fields = g_strsplit (gstclock, " ", 0);
6824 /* wrapped clock, not very interesting for now */
6825 if (fields[1] == NULL)
6828 /* remote IP address and port */
6829 if ((str = fields[2]) == NULL)
6832 parts = g_strsplit (str, ":", 0);
6834 if ((remote_ip = parts[0]) == NULL)
6837 if ((str = parts[1]) == NULL)
6845 if ((str = fields[3]) == NULL)
6848 base_time = g_ascii_strtoull (str, NULL, 10);
6851 gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
6854 if (src->provided_clock)
6855 gst_object_unref (src->provided_clock);
6856 src->provided_clock = netclock;
6858 gst_element_post_message (GST_ELEMENT_CAST (src),
6859 gst_message_new_clock_provide (GST_OBJECT_CAST (src),
6860 src->provided_clock, TRUE));
6864 g_strfreev (fields);
6870 /* must be called with the RTSP state lock */
6871 static GstRTSPResult
6872 gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
6878 /* prepare global stream caps properties */
6880 gst_structure_remove_all_fields (src->props);
6882 src->props = gst_structure_new_empty ("RTSPProperties");
6885 gst_sdp_message_dump (sdp);
6887 gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
6889 /* let the app inspect and change the SDP */
6890 g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
6892 gst_segment_init (&src->segment, GST_FORMAT_TIME);
6894 /* parse range for duration reporting. */
6899 range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
6903 /* keep track of the range and configure it in the segment */
6904 if (gst_rtspsrc_parse_range (src, range, &src->segment))
6908 /* parse clock information. This is GStreamer specific, a server can tell the
6909 * client what clock it is using and wrap that in a network clock. The
6910 * advantage of that is that we can slave to it. */
6912 const gchar *gstclock;
6915 gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
6916 if (gstclock == NULL)
6919 /* parse the clock and expose it in the provide_clock method */
6920 if (gst_rtspsrc_parse_gst_clock (src, gstclock))
6924 /* try to find a global control attribute. Note that a '*' means that we should
6925 * do aggregate control with the current url (so we don't do anything and
6926 * leave the current connection as is) */
6928 const gchar *control;
6931 control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
6932 if (control == NULL)
6935 /* only take fully qualified urls */
6936 if (g_str_has_prefix (control, "rtsp://"))
6940 g_free (src->conninfo.location);
6941 src->conninfo.location = g_strdup (control);
6942 /* make a connection for this, if there was a connection already, nothing
6944 if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
6945 GST_ERROR_OBJECT (src, "could not connect");
6948 /* we need to keep the control url separate from the connection url because
6949 * the rules for constructing the media control url need it */
6950 g_free (src->control);
6951 src->control = g_strdup (control);
6954 /* create streams */
6955 n_streams = gst_sdp_message_medias_len (sdp);
6956 for (i = 0; i < n_streams; i++) {
6957 gst_rtspsrc_create_stream (src, sdp, i);
6960 src->state = GST_RTSP_STATE_INIT;
6963 if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
6966 /* reset our state */
6967 src->need_range = TRUE;
6970 src->state = GST_RTSP_STATE_READY;
6977 GST_ERROR_OBJECT (src, "setup failed");
6978 gst_rtspsrc_cleanup (src);
6983 static GstRTSPResult
6984 gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
6988 GstRTSPMessage request = { 0 };
6989 GstRTSPMessage response = { 0 };
6992 gchar *respcont = NULL;
6995 src->need_redirect = FALSE;
6997 /* can't continue without a valid url */
6998 if (G_UNLIKELY (src->conninfo.url == NULL)) {
6999 res = GST_RTSP_EINVAL;
7002 src->tried_url_auth = FALSE;
7004 if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
7005 goto connect_failed;
7007 /* create OPTIONS */
7008 GST_DEBUG_OBJECT (src, "create options...");
7010 gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
7011 src->conninfo.url_str);
7013 goto create_request_failed;
7016 GST_DEBUG_OBJECT (src, "send options...");
7019 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
7022 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
7027 if (!gst_rtspsrc_parse_methods (src, &response))
7030 /* create DESCRIBE */
7031 GST_DEBUG_OBJECT (src, "create describe...");
7033 gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
7034 src->conninfo.url_str);
7036 goto create_request_failed;
7038 /* we only accept SDP for now */
7039 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
7043 GST_DEBUG_OBJECT (src, "send describe...");
7046 GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
7049 gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
7053 /* we only perform redirect for the describe, currently */
7054 if (src->need_redirect) {
7055 /* close connection, we don't have to send a TEARDOWN yet, ignore the
7057 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7059 gst_rtsp_message_unset (&request);
7060 gst_rtsp_message_unset (&response);
7066 /* it could be that the DESCRIBE method was not implemented */
7067 if (!src->methods & GST_RTSP_DESCRIBE)
7070 /* check if reply is SDP */
7071 gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
7073 /* could not be set but since the request returned OK, we assume it
7074 * was SDP, else check it. */
7076 if (g_ascii_strcasecmp (respcont, "application/sdp") != 0)
7077 goto wrong_content_type;
7080 /* get message body and parse as SDP */
7081 gst_rtsp_message_get_body (&response, &data, &size);
7082 if (data == NULL || size == 0)
7085 GST_DEBUG_OBJECT (src, "parse SDP...");
7086 gst_sdp_message_new (sdp);
7087 gst_sdp_message_parse_buffer (data, size, *sdp);
7089 /* clean up any messages */
7090 gst_rtsp_message_unset (&request);
7091 gst_rtsp_message_unset (&response);
7098 GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
7099 ("No valid RTSP URL was provided"));
7104 gchar *str = gst_rtsp_strresult (res);
7106 if (res != GST_RTSP_EINTR) {
7107 GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
7108 ("Failed to connect. (%s)", str));
7110 GST_WARNING_OBJECT (src, "connect interrupted");
7115 create_request_failed:
7117 gchar *str = gst_rtsp_strresult (res);
7119 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7120 ("Could not create request. (%s)", str));
7126 /* Don't post a message - the rtsp_send method will have
7127 * taken care of it because we passed NULL for the response code */
7132 /* error was posted */
7133 res = GST_RTSP_ERROR;
7138 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7139 ("Server does not support SDP, got %s.", respcont));
7140 res = GST_RTSP_ERROR;
7145 GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
7146 ("Server can not provide an SDP."));
7147 res = GST_RTSP_ERROR;
7152 if (src->conninfo.connection) {
7153 GST_DEBUG_OBJECT (src, "free connection");
7154 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7156 gst_rtsp_message_unset (&request);
7157 gst_rtsp_message_unset (&response);
7162 static GstRTSPResult
7163 gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
7168 GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
7170 if (src->sdp == NULL) {
7171 if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
7175 if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
7180 gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
7187 GST_WARNING_OBJECT (src, "can't get sdp");
7188 src->open_error = TRUE;
7193 GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
7194 src->open_error = TRUE;
7199 static GstRTSPResult
7200 gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
7202 GstRTSPMessage request = { 0 };
7203 GstRTSPMessage response = { 0 };
7204 GstRTSPResult res = GST_RTSP_OK;
7206 const gchar *control;
7208 GST_DEBUG_OBJECT (src, "TEARDOWN...");
7210 gst_rtspsrc_set_state (src, GST_STATE_READY);
7212 if (src->state < GST_RTSP_STATE_READY) {
7213 GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
7220 /* construct a control url */
7221 control = get_aggregate_control (src);
7223 if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
7226 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7227 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7228 const gchar *setup_url;
7229 GstRTSPConnInfo *info;
7231 /* try aggregate control first but do non-aggregate control otherwise */
7233 setup_url = control;
7234 else if ((setup_url = stream->conninfo.location) == NULL)
7237 if (src->conninfo.connection) {
7238 info = &src->conninfo;
7239 } else if (stream->conninfo.connection) {
7240 info = &stream->conninfo;
7244 if (!info->connected)
7249 gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
7251 goto create_request_failed;
7254 GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
7257 gst_rtspsrc_send (src, info->connection, &request, &response,
7261 /* FIXME, parse result? */
7262 gst_rtsp_message_unset (&request);
7263 gst_rtsp_message_unset (&response);
7266 /* early exit when we did aggregate control */
7272 /* close connections */
7273 GST_DEBUG_OBJECT (src, "closing connection...");
7274 gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
7275 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7276 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7277 gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
7281 gst_rtspsrc_cleanup (src);
7283 src->state = GST_RTSP_STATE_INVALID;
7286 gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
7291 create_request_failed:
7293 gchar *str = gst_rtsp_strresult (res);
7295 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7296 ("Could not create request. (%s)", str));
7302 gchar *str = gst_rtsp_strresult (res);
7304 gst_rtsp_message_unset (&request);
7305 if (res != GST_RTSP_EINTR) {
7306 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7307 ("Could not send message. (%s)", str));
7309 GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
7316 GST_DEBUG_OBJECT (src,
7317 "TEARDOWN and PLAY not supported, can't do TEARDOWN");
7322 /* RTP-Info is of the format:
7324 * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
7326 * rtptime corresponds to the timestamp for the NPT time given in the header
7327 * seqbase corresponds to the next sequence number we received. This number
7328 * indicates the first seqnum after the seek and should be used to discard
7329 * packets that are from before the seek.
7332 gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
7337 GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
7339 infos = g_strsplit (rtpinfo, ",", 0);
7340 for (i = 0; infos[i]; i++) {
7342 GstRTSPStream *stream;
7346 GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
7348 /* init values, types of seqbase and timebase are bigger than needed so we
7349 * can store -1 as uninitialized values */
7354 /* parse url, find stream for url.
7355 * parse seq and rtptime. The seq number should be configured in the rtp
7356 * depayloader or session manager to detect gaps. Same for the rtptime, it
7357 * should be used to create an initial time newsegment. */
7358 fields = g_strsplit (infos[i], ";", 0);
7359 for (j = 0; fields[j]; j++) {
7360 GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
7361 /* remove leading whitespace */
7362 fields[j] = g_strchug (fields[j]);
7363 if (g_str_has_prefix (fields[j], "url=")) {
7364 /* get the url and the stream */
7366 find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
7367 } else if (g_str_has_prefix (fields[j], "seq=")) {
7368 seqbase = atoi (fields[j] + 4);
7369 } else if (g_str_has_prefix (fields[j], "rtptime=")) {
7370 timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
7373 g_strfreev (fields);
7374 /* now we need to store the values for the caps of the stream */
7375 if (stream != NULL) {
7376 GST_DEBUG_OBJECT (src,
7377 "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
7378 stream, seqbase, timebase);
7380 /* we have a stream, configure detected params */
7381 stream->seqbase = seqbase;
7382 stream->timebase = timebase;
7391 gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
7396 interval = strtoul (rtcp, NULL, 10);
7397 GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
7402 interval *= GST_MSECOND;
7404 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7405 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7407 /* already (optionally) retrieved this when configuring manager */
7408 if (stream->session) {
7409 GObject *rtpsession = stream->session;
7411 GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
7413 g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
7417 /* now it happens that (Xenon) server sending this may also provide bogus
7418 * RTCP SR sync data (i.e. with quite some jitter), so never mind those
7419 * and just use RTP-Info to sync */
7421 GObjectClass *klass;
7423 klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
7424 if (g_object_class_find_property (klass, "rtcp-sync")) {
7425 GST_DEBUG_OBJECT (src, "configuring rtp sync method");
7426 g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
7432 gst_rtspsrc_get_float (const gchar * dstr)
7434 gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7436 /* canonicalise floating point string so we can handle float strings
7437 * in the form "24.930" or "24,930" irrespective of the current locale */
7438 g_strlcpy (s, dstr, sizeof (s));
7439 g_strdelimit (s, ",", '.');
7440 return g_ascii_strtod (s, NULL);
7444 gen_range_header (GstRTSPSrc * src, GstSegment * segment)
7446 gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
7448 if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
7449 g_strlcpy (val_str, "now", sizeof (val_str));
7451 if (segment->position == 0) {
7452 g_strlcpy (val_str, "0", sizeof (val_str));
7454 g_ascii_dtostr (val_str, sizeof (val_str),
7455 ((gdouble) segment->position) / GST_SECOND);
7458 return g_strdup_printf ("npt=%s-", val_str);
7462 clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
7466 stream->timebase = -1;
7467 stream->seqbase = -1;
7469 len = stream->ptmap->len;
7470 for (i = 0; i < len; i++) {
7471 PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
7474 if (item->caps == NULL)
7477 item->caps = gst_caps_make_writable (item->caps);
7478 s = gst_caps_get_structure (item->caps, 0);
7479 gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
7483 static GstRTSPResult
7484 gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
7486 GstRTSPResult res = GST_RTSP_OK;
7488 if (src->state < GST_RTSP_STATE_READY) {
7489 res = GST_RTSP_ERROR;
7490 if (src->open_error) {
7491 GST_DEBUG_OBJECT (src, "the stream was in error");
7495 gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
7497 if ((res = gst_rtspsrc_open (src, async)) < 0) {
7498 GST_DEBUG_OBJECT (src, "failed to open stream");
7507 static GstRTSPResult
7508 gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
7510 GstRTSPMessage request = { 0 };
7511 GstRTSPMessage response = { 0 };
7512 GstRTSPResult res = GST_RTSP_OK;
7516 const gchar *control;
7518 GST_DEBUG_OBJECT (src, "PLAY...");
7520 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7523 if (!(src->methods & GST_RTSP_PLAY))
7526 if (src->state == GST_RTSP_STATE_PLAYING)
7529 if (!src->conninfo.connection || !src->conninfo.connected)
7532 /* send some dummy packets before we activate the receive in the
7534 gst_rtspsrc_send_dummy_packets (src);
7536 /* require new SR packets */
7538 g_signal_emit_by_name (src->manager, "reset-sync", NULL);
7540 /* construct a control url */
7541 control = get_aggregate_control (src);
7543 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7544 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7545 const gchar *setup_url;
7546 GstRTSPConnection *conn;
7548 /* try aggregate control first but do non-aggregate control otherwise */
7550 setup_url = control;
7551 else if ((setup_url = stream->conninfo.location) == NULL)
7554 if (src->conninfo.connection) {
7555 conn = src->conninfo.connection;
7556 } else if (stream->conninfo.connection) {
7557 conn = stream->conninfo.connection;
7563 res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
7565 goto create_request_failed;
7567 if (src->need_range) {
7568 hval = gen_range_header (src, segment);
7570 gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
7572 /* store the newsegment event so it can be sent from the streaming thread. */
7573 src->need_segment = TRUE;
7576 if (segment->rate != 1.0) {
7577 gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
7579 g_ascii_dtostr (hval, sizeof (hval), segment->rate);
7581 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
7583 gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
7587 GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
7589 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7592 /* seek may have silently failed as it is not supported */
7593 if (!(src->methods & GST_RTSP_PLAY)) {
7594 GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
7595 /* obviously it is supported as we made it here */
7596 src->methods |= GST_RTSP_PLAY;
7597 src->seekable = FALSE;
7598 /* but there is nothing to parse in the response,
7599 * so convey we have no idea and not to expect anything particular */
7600 clear_rtp_base (src, stream);
7604 /* need to do for all streams */
7605 for (run = src->streams; run; run = g_list_next (run))
7606 clear_rtp_base (src, (GstRTSPStream *) run->data);
7608 /* NOTE the above also disables npt based eos detection */
7609 /* and below forces position to 0,
7610 * which is visible feedback we lost the plot */
7611 segment->start = segment->position = src->last_pos;
7614 gst_rtsp_message_unset (&request);
7616 /* parse RTP npt field. This is the current position in the stream (Normal
7617 * Play Time) and should be put in the NEWSEGMENT position field. */
7618 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
7620 gst_rtspsrc_parse_range (src, hval, segment);
7622 /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
7623 segment->rate = 1.0;
7625 /* parse Speed header. This is the intended playback rate of the stream
7626 * and should be put in the NEWSEGMENT rate field. */
7627 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
7628 0) == GST_RTSP_OK) {
7629 segment->rate = gst_rtspsrc_get_float (hval);
7630 } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
7631 &hval, 0) == GST_RTSP_OK) {
7632 segment->rate = gst_rtspsrc_get_float (hval);
7635 /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
7636 * for the RTP packets. If this is not present, we assume all starts from 0...
7637 * This is info for the RTP session manager that we pass to it in caps. */
7639 while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
7640 &hval, hval_idx++) == GST_RTSP_OK)
7641 gst_rtspsrc_parse_rtpinfo (src, hval);
7643 /* some servers indicate RTCP parameters in PLAY response,
7644 * rather than properly in SDP */
7645 if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
7646 &hval, 0) == GST_RTSP_OK)
7647 gst_rtspsrc_handle_rtcp_interval (src, hval);
7649 gst_rtsp_message_unset (&response);
7651 /* early exit when we did aggregate control */
7655 /* configure the caps of the streams after we parsed all headers. Only reset
7656 * the manager object when we set a new Range header (we did a seek) */
7657 gst_rtspsrc_configure_caps (src, segment, src->need_range);
7659 /* set to PLAYING after we have configured the caps, otherwise we
7660 * might end up calling request_key (with SRTP) while caps are still
7661 * being configured. */
7662 gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
7664 /* set again when needed */
7665 src->need_range = FALSE;
7667 src->running = TRUE;
7668 src->base_time = -1;
7669 src->state = GST_RTSP_STATE_PLAYING;
7672 GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
7673 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7674 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7675 stream->discont = TRUE;
7680 gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
7687 GST_DEBUG_OBJECT (src, "failed to open stream");
7692 GST_DEBUG_OBJECT (src, "PLAY is not supported");
7697 GST_DEBUG_OBJECT (src, "we were already PLAYING");
7700 create_request_failed:
7702 gchar *str = gst_rtsp_strresult (res);
7704 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7705 ("Could not create request. (%s)", str));
7711 gchar *str = gst_rtsp_strresult (res);
7713 gst_rtsp_message_unset (&request);
7714 if (res != GST_RTSP_EINTR) {
7715 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7716 ("Could not send message. (%s)", str));
7718 GST_WARNING_OBJECT (src, "PLAY interrupted");
7725 static GstRTSPResult
7726 gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
7728 GstRTSPResult res = GST_RTSP_OK;
7729 GstRTSPMessage request = { 0 };
7730 GstRTSPMessage response = { 0 };
7732 const gchar *control;
7734 GST_DEBUG_OBJECT (src, "PAUSE...");
7736 if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
7739 if (!(src->methods & GST_RTSP_PAUSE))
7742 if (src->state == GST_RTSP_STATE_READY)
7745 if (!src->conninfo.connection || !src->conninfo.connected)
7748 /* construct a control url */
7749 control = get_aggregate_control (src);
7751 /* loop over the streams. We might exit the loop early when we could do an
7752 * aggregate control */
7753 for (walk = src->streams; walk; walk = g_list_next (walk)) {
7754 GstRTSPStream *stream = (GstRTSPStream *) walk->data;
7755 GstRTSPConnection *conn;
7756 const gchar *setup_url;
7758 /* try aggregate control first but do non-aggregate control otherwise */
7760 setup_url = control;
7761 else if ((setup_url = stream->conninfo.location) == NULL)
7764 if (src->conninfo.connection) {
7765 conn = src->conninfo.connection;
7766 } else if (stream->conninfo.connection) {
7767 conn = stream->conninfo.connection;
7773 GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
7774 ("Sending PAUSE request"));
7777 gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
7779 goto create_request_failed;
7781 if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
7784 gst_rtsp_message_unset (&request);
7785 gst_rtsp_message_unset (&response);
7787 /* exit early when we did agregate control */
7792 /* change element states now */
7793 gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
7796 src->state = GST_RTSP_STATE_READY;
7800 gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
7807 GST_DEBUG_OBJECT (src, "failed to open stream");
7812 GST_DEBUG_OBJECT (src, "PAUSE is not supported");
7817 GST_DEBUG_OBJECT (src, "we were already PAUSED");
7820 create_request_failed:
7822 gchar *str = gst_rtsp_strresult (res);
7824 GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
7825 ("Could not create request. (%s)", str));
7831 gchar *str = gst_rtsp_strresult (res);
7833 gst_rtsp_message_unset (&request);
7834 if (res != GST_RTSP_EINTR) {
7835 GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
7836 ("Could not send message. (%s)", str));
7838 GST_WARNING_OBJECT (src, "PAUSE interrupted");
7846 gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
7848 GstRTSPSrc *rtspsrc;
7850 rtspsrc = GST_RTSPSRC (bin);
7852 switch (GST_MESSAGE_TYPE (message)) {
7853 case GST_MESSAGE_EOS:
7854 gst_message_unref (message);
7856 case GST_MESSAGE_ELEMENT:
7858 const GstStructure *s = gst_message_get_structure (message);
7860 if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
7861 gboolean ignore_timeout;
7863 GST_DEBUG_OBJECT (bin, "timeout on UDP port");
7865 GST_OBJECT_LOCK (rtspsrc);
7866 ignore_timeout = rtspsrc->ignore_timeout;
7867 rtspsrc->ignore_timeout = TRUE;
7868 GST_OBJECT_UNLOCK (rtspsrc);
7870 /* we only act on the first udp timeout message, others are irrelevant
7871 * and can be ignored. */
7872 if (!ignore_timeout)
7873 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
7875 gst_message_unref (message);
7878 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7881 case GST_MESSAGE_ERROR:
7884 GstRTSPStream *stream;
7887 udpsrc = GST_MESSAGE_SRC (message);
7889 GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
7890 GST_ELEMENT_NAME (udpsrc));
7892 stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
7896 /* we ignore the RTCP udpsrc */
7897 if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
7900 /* if we get error messages from the udp sources, that's not a problem as
7901 * long as not all of them error out. We also don't really know what the
7902 * problem is, the message does not give enough detail... */
7903 ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
7904 GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
7905 if (ret != GST_FLOW_OK)
7909 gst_message_unref (message);
7913 /* fatal but not our message, forward */
7914 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7919 GST_BIN_CLASS (parent_class)->handle_message (bin, message);
7925 /* the thread where everything happens */
7927 gst_rtspsrc_thread (GstRTSPSrc * src)
7931 GST_OBJECT_LOCK (src);
7932 cmd = src->pending_cmd;
7933 if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
7934 || cmd == CMD_LOOP || cmd == CMD_OPEN)
7935 src->pending_cmd = CMD_LOOP;
7937 src->pending_cmd = CMD_WAIT;
7938 GST_DEBUG_OBJECT (src, "got command %s", cmd_to_string (cmd));
7940 /* we got the message command, so ensure communication is possible again */
7941 gst_rtspsrc_connection_flush (src, FALSE);
7943 src->busy_cmd = cmd;
7944 GST_OBJECT_UNLOCK (src);
7948 gst_rtspsrc_open (src, TRUE);
7951 gst_rtspsrc_play (src, &src->segment, TRUE);
7954 gst_rtspsrc_pause (src, TRUE);
7957 gst_rtspsrc_close (src, TRUE, FALSE);
7960 gst_rtspsrc_loop (src);
7963 gst_rtspsrc_reconnect (src, FALSE);
7969 GST_OBJECT_LOCK (src);
7970 /* and go back to sleep */
7971 if (src->pending_cmd == CMD_WAIT) {
7973 gst_task_pause (src->task);
7976 src->busy_cmd = CMD_WAIT;
7977 GST_OBJECT_UNLOCK (src);
7981 gst_rtspsrc_start (GstRTSPSrc * src)
7983 GST_DEBUG_OBJECT (src, "starting");
7985 GST_OBJECT_LOCK (src);
7987 src->pending_cmd = CMD_WAIT;
7989 if (src->task == NULL) {
7990 src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
7991 if (src->task == NULL)
7994 gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
7996 GST_OBJECT_UNLOCK (src);
8003 GST_OBJECT_UNLOCK (src);
8004 GST_ERROR_OBJECT (src, "failed to create task");
8010 gst_rtspsrc_stop (GstRTSPSrc * src)
8014 GST_DEBUG_OBJECT (src, "stopping");
8016 /* also cancels pending task */
8017 gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
8019 GST_OBJECT_LOCK (src);
8020 if ((task = src->task)) {
8022 GST_OBJECT_UNLOCK (src);
8024 gst_task_stop (task);
8026 /* make sure it is not running */
8027 GST_RTSP_STREAM_LOCK (src);
8028 GST_RTSP_STREAM_UNLOCK (src);
8030 /* now wait for the task to finish */
8031 gst_task_join (task);
8033 /* and free the task */
8034 gst_object_unref (GST_OBJECT (task));
8036 GST_OBJECT_LOCK (src);
8038 GST_OBJECT_UNLOCK (src);
8040 /* ensure synchronously all is closed and clean */
8041 gst_rtspsrc_close (src, FALSE, TRUE);
8046 static GstStateChangeReturn
8047 gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
8049 GstRTSPSrc *rtspsrc;
8050 GstStateChangeReturn ret;
8052 rtspsrc = GST_RTSPSRC (element);
8054 switch (transition) {
8055 case GST_STATE_CHANGE_NULL_TO_READY:
8056 if (!gst_rtspsrc_start (rtspsrc))
8059 case GST_STATE_CHANGE_READY_TO_PAUSED:
8060 /* init some state */
8061 rtspsrc->cur_protocols = rtspsrc->protocols;
8062 /* first attempt, don't ignore timeouts */
8063 rtspsrc->ignore_timeout = FALSE;
8064 rtspsrc->open_error = FALSE;
8065 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
8067 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8068 set_manager_buffer_mode (rtspsrc);
8070 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8071 /* unblock the tcp tasks and make the loop waiting */
8072 if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
8073 /* make sure it is waiting before we send PAUSE or PLAY below */
8074 GST_RTSP_STREAM_LOCK (rtspsrc);
8075 GST_RTSP_STREAM_UNLOCK (rtspsrc);
8078 case GST_STATE_CHANGE_PAUSED_TO_READY:
8084 ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
8085 if (ret == GST_STATE_CHANGE_FAILURE)
8088 switch (transition) {
8089 case GST_STATE_CHANGE_NULL_TO_READY:
8090 ret = GST_STATE_CHANGE_SUCCESS;
8092 case GST_STATE_CHANGE_READY_TO_PAUSED:
8093 ret = GST_STATE_CHANGE_NO_PREROLL;
8095 case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
8096 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
8097 ret = GST_STATE_CHANGE_SUCCESS;
8099 case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
8100 /* send pause request and keep the idle task around */
8101 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
8102 ret = GST_STATE_CHANGE_NO_PREROLL;
8104 case GST_STATE_CHANGE_PAUSED_TO_READY:
8105 gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
8106 ret = GST_STATE_CHANGE_SUCCESS;
8108 case GST_STATE_CHANGE_READY_TO_NULL:
8109 gst_rtspsrc_stop (rtspsrc);
8110 ret = GST_STATE_CHANGE_SUCCESS;
8121 GST_DEBUG_OBJECT (rtspsrc, "start failed");
8122 return GST_STATE_CHANGE_FAILURE;
8127 gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
8130 GstRTSPSrc *rtspsrc;
8132 rtspsrc = GST_RTSPSRC (element);
8134 if (GST_EVENT_IS_DOWNSTREAM (event)) {
8135 res = gst_rtspsrc_push_event (rtspsrc, event);
8137 res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
8144 /*** GSTURIHANDLER INTERFACE *************************************************/
8147 gst_rtspsrc_uri_get_type (GType type)
8152 static const gchar *const *
8153 gst_rtspsrc_uri_get_protocols (GType type)
8155 static const gchar *protocols[] =
8156 { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
8157 "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
8164 gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
8166 GstRTSPSrc *src = GST_RTSPSRC (handler);
8168 /* FIXME: make thread-safe */
8169 return g_strdup (src->conninfo.location);
8173 gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
8179 GstRTSPUrl *newurl = NULL;
8180 GstSDPMessage *sdp = NULL;
8182 src = GST_RTSPSRC (handler);
8184 /* same URI, we're fine */
8185 if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
8188 if (g_str_has_prefix (uri, "rtsp-sdp://")) {
8189 sres = gst_sdp_message_new (&sdp);
8193 GST_DEBUG_OBJECT (src, "parsing SDP message");
8194 sres = gst_sdp_message_parse_uri (uri, sdp);
8199 GST_DEBUG_OBJECT (src, "parsing URI");
8200 if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
8204 /* if worked, free previous and store new url object along with the original
8206 GST_DEBUG_OBJECT (src, "configuring URI");
8207 g_free (src->conninfo.location);
8208 src->conninfo.location = g_strdup (uri);
8209 gst_rtsp_url_free (src->conninfo.url);
8210 src->conninfo.url = newurl;
8211 g_free (src->conninfo.url_str);
8213 src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
8215 src->conninfo.url_str = NULL;
8218 gst_sdp_message_free (src->sdp);
8220 src->from_sdp = sdp != NULL;
8222 GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
8223 GST_DEBUG_OBJECT (src, "request uri is: %s",
8224 GST_STR_NULL (src->conninfo.url_str));
8231 GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
8236 GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", sres);
8237 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8238 "Could not create SDP");
8243 GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", sres,
8244 GST_STR_NULL (uri));
8245 gst_sdp_message_free (sdp);
8246 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8252 GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
8253 GST_STR_NULL (uri), res);
8254 g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
8255 "Invalid RTSP URI");
8261 gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
8263 GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
8265 iface->get_type = gst_rtspsrc_uri_get_type;
8266 iface->get_protocols = gst_rtspsrc_uri_get_protocols;
8267 iface->get_uri = gst_rtspsrc_uri_get_uri;
8268 iface->set_uri = gst_rtspsrc_uri_set_uri;