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gstreamer/gst/rtsp/gstrtspsrc.c
Wim Taymans b14738fb20 gst/rtsp/: Reuse already existing enum for lower transport. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/rtspconnection.c: (rtsp_connection_create):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
* gst/rtsp/rtspurl.h:
Reuse already existing enum for lower transport.
Add rtspt and rtspu protocols.
Send redirect to rtspt when udp times out.
2006-10-18 16:18:55 +00:00

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/* GStreamer
* Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
* <2006> Lutz Mueller <lutz at topfrose dot de>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
* Unless otherwise indicated, Source Code is licensed under MIT license.
* See further explanation attached in License Statement (distributed in the file
* LICENSE).
*
* Permission is hereby granted, free of charge, to any person obtaining a copy of
* this software and associated documentation files (the "Software"), to deal in
* the Software without restriction, including without limitation the rights to
* use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
* of the Software, and to permit persons to whom the Software is furnished to do
* so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in all
* copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
* SOFTWARE.
*/
/**
* SECTION:element-rtspsrc
*
* <refsect2>
* <para>
* Makes a connection to an RTSP server and read the data.
* rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
* RealMedia/Quicktime/Microsoft extensions.
* </para>
* <para>
* RTSP supports transport over TCP or UDP in unicast or multicast mode. By
* default rtspsrc will negotiate a connection in the following order:
* UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
* protocols can be controlled with the "protocols" property.
* </para>
* <para>
* rtspsrc currently understands SDP as the format of the session description.
* For each stream listed in the SDP a new rtp_stream%d pad will be created
* with caps derived from the SDP media description. This is a caps of mime type
* "application/x-rtp" that can be connected to any available RTP depayloader
* element.
* </para>
* <para>
* rtspsrc will internally instantiate an RTP session manager element
* that will handle the RTCP messages to and from the server, jitter removal,
* packet reordering along with providing a clock for the pipeline.
* This feature is however currently not yet implemented.
* </para>
* <para>
* rtspsrc acts like a live source and will therefore only generate data in the
* PLAYING state.
* </para>
* <title>Example launch line</title>
* <para>
* <programlisting>
* gst-launch rtspsrc location=rtsp://some.server/url ! fakesink
* </programlisting>
* Establish a connection to an RTSP server and send the raw RTP packets to a fakesink.
* </para>
* </refsect2>
*
* Last reviewed on 2006-08-18 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <unistd.h>
#include <stdlib.h>
#include <string.h>
#include "gstrtspsrc.h"
#include "sdp.h"
#if 0
#define WITH_EXT_REAL
#endif
#include "rtspextwms.h"
#ifdef WITH_EXT_REAL
#include "rtspextreal.h"
#endif
GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
#define GST_CAT_DEFAULT (rtspsrc_debug)
/* elementfactory information */
static const GstElementDetails gst_rtspsrc_details =
GST_ELEMENT_DETAILS ("RTSP packet receiver",
"Source/Network",
"Receive data over the network via RTSP (RFC 2326)",
"Wim Taymans <wim@fluendo.com>\n"
"Thijs Vermeir <thijs.vermeir@barco.com>\n"
"Lutz Mueller <lutz@topfrose.de>");
static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_LOCATION NULL
#define DEFAULT_PROTOCOLS RTSP_LOWER_TRANS_UDP | RTSP_LOWER_TRANS_UDP_MCAST | RTSP_LOWER_TRANS_TCP
#define DEFAULT_DEBUG FALSE
#define DEFAULT_RETRY 20
#define DEFAULT_TIMEOUT 5000000
enum
{
PROP_0,
PROP_LOCATION,
PROP_PROTOCOLS,
PROP_DEBUG,
PROP_RETRY,
PROP_TIMEOUT,
};
#define GST_TYPE_RTSP_LOWER_TRANS (gst_rtsp_lower_trans_get_type())
static GType
gst_rtsp_lower_trans_get_type (void)
{
static GType rtsp_lower_trans_type = 0;
static const GFlagsValue rtsp_lower_trans[] = {
{RTSP_LOWER_TRANS_UDP, "UDP Unicast Mode", "udp-unicast"},
{RTSP_LOWER_TRANS_UDP_MCAST, "UDP Multicast Mode", "udp-multicast"},
{RTSP_LOWER_TRANS_TCP, "TCP interleaved mode", "tcp"},
{0, NULL, NULL},
};
if (!rtsp_lower_trans_type) {
rtsp_lower_trans_type =
g_flags_register_static ("GstRTSPLowerTrans", rtsp_lower_trans);
}
return rtsp_lower_trans_type;
}
static void gst_rtspsrc_base_init (gpointer g_class);
static void gst_rtspsrc_finalize (GObject * object);
static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static GstCaps *gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media);
static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
GstStateChange transition);
static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rtspsrc_open (GstRTSPSrc * src);
static gboolean gst_rtspsrc_play (GstRTSPSrc * src);
static gboolean gst_rtspsrc_pause (GstRTSPSrc * src);
static gboolean gst_rtspsrc_close (GstRTSPSrc * src);
static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
const gchar * uri);
static void gst_rtspsrc_loop (GstRTSPSrc * src);
/* commands we send to out loop to notify it of events */
#define CMD_WAIT 0
#define CMD_RECONNECT 1
#define CMD_STOP 2
/*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
static void
_do_init (GType rtspsrc_type)
{
static const GInterfaceInfo urihandler_info = {
gst_rtspsrc_uri_handler_init,
NULL,
NULL
};
GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
&urihandler_info);
}
GST_BOILERPLATE_FULL (GstRTSPSrc, gst_rtspsrc, GstBin, GST_TYPE_BIN, _do_init);
static void
gst_rtspsrc_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtptemplate));
gst_element_class_set_details (element_class, &gst_rtspsrc_details);
}
static void
gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBinClass *gstbin_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbin_class = (GstBinClass *) klass;
gobject_class->set_property = gst_rtspsrc_set_property;
gobject_class->get_property = gst_rtspsrc_get_property;
gobject_class->finalize = gst_rtspsrc_finalize;
g_object_class_install_property (gobject_class, PROP_LOCATION,
g_param_spec_string ("location", "RTSP Location",
"Location of the RTSP url to read",
DEFAULT_LOCATION, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols",
"Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_DEBUG,
g_param_spec_boolean ("debug", "Debug",
"Dump request and response messages to stdout",
DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_RETRY,
g_param_spec_uint ("retry", "Retry",
"Max number of retries when allocating RTP ports.",
0, G_MAXUINT16, DEFAULT_RETRY,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_TIMEOUT,
g_param_spec_uint64 ("timeout", "Timeout",
"Retry TCP transport after timeout microseconds (0 = disabled)",
0, G_MAXUINT64, DEFAULT_TIMEOUT,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT));
gstelement_class->change_state = gst_rtspsrc_change_state;
gstbin_class->handle_message = gst_rtspsrc_handle_message;
}
static void
gst_rtspsrc_init (GstRTSPSrc * src, GstRTSPSrcClass * g_class)
{
src->stream_rec_lock = g_new (GStaticRecMutex, 1);
g_static_rec_mutex_init (src->stream_rec_lock);
src->loop_cond = g_cond_new ();
src->location = g_strdup (DEFAULT_LOCATION);
src->url = NULL;
/* install WMS extension by default */
src->extension = rtsp_ext_wms_get_context ();
#ifdef WITH_EXT_REAL
src->extension = rtsp_ext_real_get_context ();
#endif
src->extension->src = (gpointer) src;
}
static void
gst_rtspsrc_finalize (GObject * object)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
g_static_rec_mutex_free (rtspsrc->stream_rec_lock);
g_free (rtspsrc->stream_rec_lock);
g_cond_free (rtspsrc->loop_cond);
g_free (rtspsrc->location);
g_free (rtspsrc->content_base);
rtsp_url_free (rtspsrc->url);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
GParamSpec * pspec)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
switch (prop_id) {
case PROP_LOCATION:
gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
g_value_get_string (value));
break;
case PROP_PROTOCOLS:
rtspsrc->protocols = g_value_get_flags (value);
break;
case PROP_DEBUG:
rtspsrc->debug = g_value_get_boolean (value);
break;
case PROP_RETRY:
rtspsrc->retry = g_value_get_uint (value);
break;
case PROP_TIMEOUT:
rtspsrc->timeout = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRTSPSrc *rtspsrc;
rtspsrc = GST_RTSPSRC (object);
switch (prop_id) {
case PROP_LOCATION:
g_value_set_string (value, rtspsrc->location);
break;
case PROP_PROTOCOLS:
g_value_set_flags (value, rtspsrc->protocols);
break;
case PROP_DEBUG:
g_value_set_boolean (value, rtspsrc->debug);
break;
case PROP_RETRY:
g_value_set_uint (value, rtspsrc->retry);
break;
case PROP_TIMEOUT:
g_value_set_uint64 (value, rtspsrc->timeout);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gint
find_stream_by_pt (GstRTSPStream * stream, gconstpointer a)
{
gint pt = GPOINTER_TO_INT (a);
if (stream->pt == pt)
return 0;
return -1;
}
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src, SDPMessage * sdp, gint idx)
{
GstRTSPStream *stream;
gchar *control_url;
gchar *payload;
SDPMedia *media;
/* get media, should not return NULL */
media = sdp_message_get_media (sdp, idx);
if (media == NULL)
return NULL;
stream = g_new0 (GstRTSPStream, 1);
stream->parent = src;
/* we mark the pad as not linked, we will mark it as OK when we add the pad to
* the element. */
stream->last_ret = GST_FLOW_NOT_LINKED;
stream->id = src->numstreams++;
/* we must have a payload. No payload means we cannot create caps */
/* FIXME, handle multiple formats. */
if ((payload = sdp_media_get_format (media, 0))) {
stream->pt = atoi (payload);
/* convert caps */
stream->caps = gst_rtspsrc_media_to_caps (stream->pt, media);
if (stream->pt >= 96) {
/* If we have a dynamic payload type, see if we have a stream with the
* same payload number. If there is one, they are part of the same
* container and we only need to add one pad. */
if (g_list_find_custom (src->streams, GINT_TO_POINTER (stream->pt),
(GCompareFunc) find_stream_by_pt)) {
stream->container = TRUE;
}
}
}
/* get control url to construct the setup url. The setup url is used to
* configure the transport of the stream and is used to identity the stream in
* the RTP-Info header field returned from PLAY. */
control_url = sdp_media_get_attribute_val (media, "control");
GST_DEBUG_OBJECT (src, "stream %d", stream->id);
GST_DEBUG_OBJECT (src, " pt: %d", stream->pt);
GST_DEBUG_OBJECT (src, " container: %d", stream->container);
GST_DEBUG_OBJECT (src, " caps: %" GST_PTR_FORMAT, stream->caps);
GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
if (control_url != NULL) {
/* If the control_url starts with a '/' or a non rtsp: protocol we will most
* likely build a URL that the server will fail to understand, this is ok,
* we will fail then. */
if (g_str_has_prefix (control_url, "rtsp://"))
stream->setup_url = g_strdup (control_url);
else if (src->content_base)
stream->setup_url =
g_strdup_printf ("%s%s", src->content_base, control_url);
else
stream->setup_url = g_strdup_printf ("%s/%s", src->location, control_url);
}
GST_DEBUG_OBJECT (src, " setup: %s", GST_STR_NULL (stream->setup_url));
/* we keep track of all streams */
src->streams = g_list_append (src->streams, stream);
return stream;
}
#if 0
static void
gst_rtspsrc_free_stream (GstRTSPSrc * src, GstRTSPStream * stream)
{
if (stream->caps) {
gst_caps_unref (stream->caps);
}
g_free (stream->setup_url);
src->streams = g_list_remove (src->streams, stream);
src->numstreams--;
g_free (stream);
}
#endif
#define PARSE_INT(p, del, res) \
G_STMT_START { \
gchar *t = p; \
p = strstr (p, del); \
if (p == NULL) \
res = -1; \
else { \
*p = '\0'; \
p++; \
res = atoi (t); \
} \
} G_STMT_END
#define PARSE_STRING(p, del, res) \
G_STMT_START { \
gchar *t = p; \
p = strstr (p, del); \
if (p == NULL) \
res = NULL; \
else { \
*p = '\0'; \
p++; \
res = t; \
} \
} G_STMT_END
#define SKIP_SPACES(p) \
while (*p && g_ascii_isspace (*p)) \
p++;
/* rtpmap contains:
*
* <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
*/
static gboolean
gst_rtspsrc_parse_rtpmap (gchar * rtpmap, gint * payload, gchar ** name,
gint * rate, gchar ** params)
{
gchar *p, *t;
t = p = rtpmap;
PARSE_INT (p, " ", *payload);
if (*payload == -1)
return FALSE;
SKIP_SPACES (p);
if (*p == '\0')
return FALSE;
PARSE_STRING (p, "/", *name);
if (*name == NULL) {
/* no rate, assume 0 then */
*name = p;
*rate = -1;
return TRUE;
}
t = p;
p = strstr (p, "/");
if (p == NULL) {
*rate = atoi (t);
return TRUE;
}
*p = '\0';
p++;
*rate = atoi (t);
t = p;
if (*p == '\0')
return TRUE;
*params = t;
return TRUE;
}
/*
* Mapping of caps to and from SDP fields:
*
* m=<media> <UDP port> RTP/AVP <payload>
* a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
* a=fmtp:<payload> <param>[=<value>];...
*/
static GstCaps *
gst_rtspsrc_media_to_caps (gint pt, SDPMedia * media)
{
GstCaps *caps;
gchar *rtpmap;
gchar *fmtp;
gchar *name = NULL;
gint rate = -1;
gchar *params = NULL;
GstStructure *s;
/* dynamic payloads need rtpmap */
if (pt >= 96) {
gint payload = 0;
gboolean ret;
if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) {
ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
if (ret) {
if (payload != pt) {
/* FIXME, not fatal? */
g_warning ("rtpmap of wrong payload type");
name = NULL;
rate = -1;
params = NULL;
}
} else {
/* FIXME, not fatal? */
g_warning ("error parsing rtpmap");
}
} else
goto no_rtpmap;
}
caps = gst_caps_new_simple ("application/x-unknown",
"media", G_TYPE_STRING, media->media, "payload", G_TYPE_INT, pt, NULL);
s = gst_caps_get_structure (caps, 0);
if (rate != -1)
gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
if (name != NULL)
gst_structure_set (s, "encoding-name", G_TYPE_STRING, name, NULL);
if (params != NULL)
gst_structure_set (s, "encoding-params", G_TYPE_STRING, params, NULL);
/* parse optional fmtp: field */
if ((fmtp = sdp_media_get_attribute_val (media, "fmtp"))) {
gchar *p;
gint payload = 0;
p = fmtp;
/* p is now of the format <payload> <param>[=<value>];... */
PARSE_INT (p, " ", payload);
if (payload != -1 && payload == pt) {
gchar **pairs;
gint i;
/* <param>[=<value>] are separated with ';' */
pairs = g_strsplit (p, ";", 0);
for (i = 0; pairs[i]; i++) {
gchar *valpos;
gchar *val, *key;
/* the key may not have a '=', the value can have other '='s */
valpos = strstr (pairs[i], "=");
if (valpos) {
/* we have a '=' and thus a value, remove the '=' with \0 */
*valpos = '\0';
/* value is everything between '=' and ';'. FIXME, strip? */
val = g_strstrip (valpos + 1);
} else {
/* simple <param>;.. is translated into <param>=1;... */
val = "1";
}
/* strip the key of spaces */
key = g_strstrip (pairs[i]);
gst_structure_set (s, key, G_TYPE_STRING, val, NULL);
}
g_strfreev (pairs);
}
}
return caps;
/* ERRORS */
no_rtpmap:
{
g_warning ("rtpmap type not given for dynamic payload %d", pt);
return NULL;
}
}
static gboolean
gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
gint * rtpport, gint * rtcpport)
{
GstRTSPSrc *src;
GstStateChangeReturn ret;
GstElement *tmp, *udpsrc0, *udpsrc1;
gint tmp_rtp, tmp_rtcp;
guint count;
src = stream->parent;
tmp = NULL;
udpsrc0 = NULL;
udpsrc1 = NULL;
count = 0;
/* try to allocate 2 UDP ports, the RTP port should be an even
* number and the RTCP port should be the next (uneven) port */
again:
udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
if (udpsrc0 == NULL)
goto no_udp_protocol;
ret = gst_element_set_state (udpsrc0, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_udp_failure;
g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
/* check if port is even */
if ((tmp_rtp & 0x01) != 0) {
/* port not even, close and allocate another */
count++;
if (count > src->retry)
goto no_ports;
GST_DEBUG_OBJECT (src, "RTP port not even, retry %d", count);
/* have to keep port allocated so we can get a new one */
if (tmp != NULL) {
GST_DEBUG_OBJECT (src, "free temp");
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
tmp = udpsrc0;
GST_DEBUG_OBJECT (src, "retry %d", count);
goto again;
}
/* free leftover temp element/port */
if (tmp) {
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
tmp = NULL;
}
/* allocate port+1 for RTCP now */
udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
if (udpsrc1 == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, NULL);
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
ret = gst_element_set_state (udpsrc1, GST_STATE_PAUSED);
/* FIXME, this could fail if the next port is not free, we
* should retry with another port then */
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtcp_failure;
/* all fine, do port check */
g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
/* this should not happen */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
/* we keep these elements, we configure all in configure_transport when the
* server told us to really use the UDP ports. */
stream->udpsrc[0] = udpsrc0;
stream->udpsrc[1] = udpsrc1;
return TRUE;
/* ERRORS */
no_udp_protocol:
{
GST_DEBUG_OBJECT (src, "could not get UDP source");
goto cleanup;
}
start_udp_failure:
{
GST_DEBUG_OBJECT (src, "could not start UDP source");
goto cleanup;
}
no_ports:
{
GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
count);
goto cleanup;
}
no_udp_rtcp_protocol:
{
GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
goto cleanup;
}
start_rtcp_failure:
{
GST_DEBUG_OBJECT (src, "could not start UDP source for RTCP");
goto cleanup;
}
port_error:
{
GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
goto cleanup;
}
cleanup:
{
if (tmp) {
gst_element_set_state (tmp, GST_STATE_NULL);
gst_object_unref (tmp);
}
if (udpsrc0) {
gst_element_set_state (udpsrc0, GST_STATE_NULL);
gst_object_unref (udpsrc0);
}
if (udpsrc1) {
gst_element_set_state (udpsrc1, GST_STATE_NULL);
gst_object_unref (udpsrc1);
}
return FALSE;
}
}
static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
RTSPTransport * transport)
{
GstRTSPSrc *src;
GstPad *outpad = NULL;
GstPadTemplate *template;
GstStateChangeReturn ret;
gchar *name;
GstStructure *s;
const gchar *mime, *manager;
RTSPResult res;
src = stream->parent;
GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
s = gst_caps_get_structure (stream->caps, 0);
if ((res = rtsp_transport_get_mime (transport->trans, &mime)) < 0)
goto no_mime;
if (!mime)
goto no_mime;
GST_DEBUG_OBJECT (src, "setting mime to %s", mime);
/* configure the final mime type */
gst_structure_set_name (s, mime);
if ((res = rtsp_transport_get_manager (transport->trans, &manager)) < 0)
goto no_manager;
if (manager) {
GST_DEBUG_OBJECT (src, "using manager %s", manager);
/* FIXME, the session manager needs to be shared with all the streams */
if (!(stream->sess = gst_element_factory_make (manager, NULL)))
goto no_element;
/* we manage this element */
gst_bin_add (GST_BIN_CAST (src), stream->sess);
ret = gst_element_set_state (stream->sess, GST_STATE_PAUSED);
if (ret != GST_STATE_CHANGE_SUCCESS)
goto start_session_failure;
stream->channelpad[0] = gst_element_get_pad (stream->sess, "sinkrtp");
stream->channelpad[1] = gst_element_get_pad (stream->sess, "sinkrtcp");
}
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
gint i;
/* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the
* session manager. */
stream->channel[0] = transport->interleaved.min;
stream->channel[1] = transport->interleaved.max;
GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
stream->channel[0], stream->channel[1]);
/* we can remove the allocated UDP ports now */
for (i = 0; i < 2; i++) {
if (stream->udpsrc[i]) {
gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
gst_object_unref (stream->udpsrc[i]);
stream->udpsrc[i] = NULL;
}
}
/* no session manager, send data to srcpad directly */
if (!stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "no manager, creating pad");
name = g_strdup_printf ("stream%d", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
stream->srcpad = gst_pad_new_from_template (template, name);
gst_object_unref (template);
g_free (name);
gst_pad_use_fixed_caps (stream->srcpad);
gst_pad_set_caps (stream->srcpad, stream->caps);
stream->channelpad[0] = gst_object_ref (stream->srcpad);
gst_pad_set_active (stream->srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
} else {
GST_DEBUG_OBJECT (src, "using manager source pad");
outpad = gst_element_get_pad (stream->sess, "srcrtp");
}
} else {
/* multicast was selected, create UDP sources and join the multicast
* group. */
if (transport->lower_transport == RTSP_LOWER_TRANS_UDP_MCAST) {
gchar *uri;
GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
/* creating UDP source */
if (transport->port.min != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.min);
stream->udpsrc[0] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[0] == NULL)
goto no_element;
/* change state */
gst_element_set_state (stream->udpsrc[0], GST_STATE_PAUSED);
}
/* creating another UDP source */
if (transport->port.max != -1) {
uri = g_strdup_printf ("udp://%s:%d", transport->destination,
transport->port.max);
stream->udpsrc[1] = gst_element_make_from_uri (GST_URI_SRC, uri, NULL);
g_free (uri);
if (stream->udpsrc[1] == NULL)
goto no_element;
gst_element_set_state (stream->udpsrc[1], GST_STATE_PAUSED);
}
}
/* we manage the UDP elements now. For unicast, the UDP sources where
* allocated in the stream when we suggested a transport. */
if (stream->udpsrc[0]) {
GstPad *pad;
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
GST_DEBUG_OBJECT (src, "setting up UDP source");
/* set caps */
g_object_set (G_OBJECT (stream->udpsrc[0]), "caps", stream->caps, NULL);
/* configure a timeout on the UDP port */
g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", src->timeout,
NULL);
if (stream->channelpad[0]) {
GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
/* configure for UDP delivery, we need to connect the UDP pads to
* the session plugin. */
pad = gst_element_get_pad (stream->udpsrc[0], "src");
gst_pad_link (pad, stream->channelpad[0]);
gst_object_unref (pad);
outpad = gst_element_get_pad (stream->sess, "srcrtp");
} else {
GST_DEBUG_OBJECT (src, "using UDP src pad as output");
outpad = gst_element_get_pad (stream->udpsrc[0], "src");
}
}
if (stream->udpsrc[1]) {
gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
if (stream->channelpad[1]) {
GstPad *pad;
GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
pad = gst_element_get_pad (stream->udpsrc[1], "src");
gst_pad_link (pad, stream->channelpad[1]);
gst_object_unref (pad);
}
}
}
if (outpad && !stream->srcpad) {
GST_DEBUG_OBJECT (src, "creating ghostpad");
gst_pad_use_fixed_caps (outpad);
gst_pad_set_caps (outpad, stream->caps);
/* create ghostpad */
name = g_strdup_printf ("stream%d", stream->id);
template = gst_static_pad_template_get (&rtptemplate);
stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
gst_object_unref (template);
g_free (name);
gst_object_unref (outpad);
/* and add */
gst_pad_set_active (stream->srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
}
/* mark pad as ok */
stream->last_ret = GST_FLOW_OK;
return TRUE;
/* ERRORS */
no_mime:
{
GST_DEBUG_OBJECT (src, "unknown transport");
return FALSE;
}
no_manager:
{
GST_DEBUG_OBJECT (src, "cannot get a session manager");
return FALSE;
}
no_element:
{
GST_DEBUG_OBJECT (src, "no rtpdec element found");
return FALSE;
}
start_session_failure:
{
GST_DEBUG_OBJECT (src, "could not start session");
return FALSE;
}
}
static gint
find_stream_by_channel (GstRTSPStream * stream, gconstpointer a)
{
gint channel = GPOINTER_TO_INT (a);
if (stream->channel[0] == channel || stream->channel[1] == channel)
return 0;
return -1;
}
static GstFlowReturn
gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
GstFlowReturn ret)
{
GList *streams;
/* store the value */
stream->last_ret = ret;
/* if it's success we can return the value right away */
if (GST_FLOW_IS_SUCCESS (ret))
goto done;
/* any other error that is not-linked can be returned right
* away */
if (ret != GST_FLOW_NOT_LINKED)
goto done;
/* only return NOT_LINKED if all other pads returned NOT_LINKED */
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
ret = ostream->last_ret;
/* some other return value (must be SUCCESS but we can return
* other values as well) */
if (ret != GST_FLOW_NOT_LINKED)
goto done;
}
/* if we get here, all other pads were unlinked and we return
* NOT_LINKED then */
done:
return ret;
}
static void
gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
{
GList *streams;
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
/* only pads that have a connection to the outside world */
if (ostream->srcpad == NULL)
continue;
if (ostream->channelpad[0]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (ostream->channelpad[0]))
gst_pad_push_event (ostream->channelpad[0], event);
else
gst_pad_send_event (ostream->channelpad[0], event);
}
if (ostream->channelpad[1]) {
gst_event_ref (event);
if (GST_PAD_IS_SRC (ostream->channelpad[1]))
gst_pad_push_event (ostream->channelpad[1], event);
else
gst_pad_send_event (ostream->channelpad[1], event);
}
}
gst_event_unref (event);
}
static void
gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
{
RTSPMessage response = { 0 };
RTSPResult res;
gint channel;
GList *lstream;
GstRTSPStream *stream;
GstPad *outpad = NULL;
guint8 *data;
guint size;
GstFlowReturn ret = GST_FLOW_OK;
GstCaps *caps = NULL;
GstBuffer *buf;
do {
GST_DEBUG_OBJECT (src, "doing receive");
if ((res = rtsp_connection_receive (src->connection, &response)) < 0)
goto receive_error;
GST_DEBUG_OBJECT (src, "got packet type %d", response.type);
}
while (response.type != RTSP_MESSAGE_DATA);
channel = response.type_data.data.channel;
lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (channel),
(GCompareFunc) find_stream_by_channel);
if (!lstream)
goto unknown_stream;
stream = (GstRTSPStream *) lstream->data;
if (channel == stream->channel[0]) {
outpad = stream->channelpad[0];
caps = stream->caps;
} else if (channel == stream->channel[1]) {
outpad = stream->channelpad[1];
}
/* take a look at the body to figure out what we have */
rtsp_message_get_body (&response, &data, &size);
if (size < 2)
goto invalid_length;
/* channels are not correct on some servers, do extra check */
if (data[1] >= 200 && data[1] <= 204) {
/* hmm RTCP message switch to the RTCP pad of the same stream. */
outpad = stream->channelpad[1];
}
/* we have no clue what this is, just ignore then. */
if (outpad == NULL)
goto unknown_stream;
/* and chain buffer to internal element */
rtsp_message_steal_body (&response, &data, &size);
/* strip the trailing \0 */
size -= 1;
buf = gst_buffer_new ();
GST_BUFFER_DATA (buf) = data;
GST_BUFFER_MALLOCDATA (buf) = data;
GST_BUFFER_SIZE (buf) = size;
/* don't need message anymore */
rtsp_message_unset (&response);
if (caps)
gst_buffer_set_caps (buf, caps);
GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
channel);
/* chain to the peer pad */
if (GST_PAD_IS_SINK (outpad))
ret = gst_pad_chain (outpad, buf);
else
ret = gst_pad_push (outpad, buf);
/* combine all stream flows */
ret = gst_rtspsrc_combine_flows (src, stream, ret);
if (ret != GST_FLOW_OK)
goto need_pause;
return;
/* ERRORS */
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
rtsp_message_unset (&response);
return;
}
receive_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
if (src->debug)
rtsp_message_dump (&response);
rtsp_message_unset (&response);
ret = GST_FLOW_UNEXPECTED;
goto need_pause;
}
invalid_length:
{
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Short message received."));
rtsp_message_unset (&response);
return;
}
need_pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
src->running = FALSE;
gst_task_pause (src->task);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
if (ret == GST_FLOW_UNEXPECTED) {
/* perform EOS logic */
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_segment_done (GST_OBJECT_CAST (src),
src->segment.format, src->segment.last_stop));
} else {
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
} else {
/* for fatal errors we post an error message, post the error
* first so the app knows about the error first. */
GST_ELEMENT_ERROR (src, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused, reason %s (%d)", reason, ret));
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
}
return;
}
}
static void
gst_rtspsrc_loop_udp (GstRTSPSrc * src)
{
gboolean restart = FALSE;
GST_OBJECT_LOCK (src);
if (src->loop_cmd == CMD_STOP)
goto stopping;
while (src->loop_cmd == CMD_WAIT) {
GST_DEBUG_OBJECT (src, "waiting");
GST_RTSP_LOOP_WAIT (src);
GST_DEBUG_OBJECT (src, "waiting done");
if (src->loop_cmd == CMD_STOP)
goto stopping;
}
if (src->loop_cmd == CMD_RECONNECT) {
/* FIXME, when we get here we have to reconnect using tcp */
src->loop_cmd = CMD_WAIT;
restart = TRUE;
}
GST_OBJECT_UNLOCK (src);
if (restart) {
gst_rtspsrc_pause (src);
if (src->task) {
/* stop task, we cannot join as this would deadlock */
gst_task_stop (src->task);
/* and free the task so that _close will not stop/join it again. */
gst_object_unref (GST_OBJECT (src->task));
src->task = NULL;
}
gst_rtspsrc_close (src);
/* see if we have TCP left to try */
if (src->cur_protocols & RTSP_LOWER_TRANS_TCP) {
gchar *url, *pos;
/* We post a warning message now to inform the user
* that nothing happened. It's most likely a firewall thing. */
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. Retrying using a TCP connection.",
(gdouble) src->timeout / 1000000));
/* we can try only TCP now */
src->cur_protocols = RTSP_LOWER_TRANS_TCP;
pos = strstr (src->location, "://");
if (!pos)
goto weird_url;
url = g_strdup_printf ("rtspt://%s", pos + 3);
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_element (GST_OBJECT_CAST (src),
gst_structure_new ("redirect",
"new-location", G_TYPE_STRING, url, NULL)));
g_free (url);
} else {
src->cur_protocols = 0;
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not connect to server, no protocols left"));
}
}
return;
/* ERRORS */
stopping:
{
GST_OBJECT_UNLOCK (src);
src->running = FALSE;
gst_task_pause (src->task);
return;
}
weird_url:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not redirect, location %s is invalid", src->location));
return;
}
}
static void
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd)
{
GST_OBJECT_LOCK (src);
src->loop_cmd = cmd;
GST_RTSP_LOOP_SIGNAL (src);
GST_OBJECT_UNLOCK (src);
}
static void
gst_rtspsrc_loop (GstRTSPSrc * src)
{
if (src->interleaved)
gst_rtspsrc_loop_interleaved (src);
else
gst_rtspsrc_loop_udp (src);
}
static RTSPResult
gst_rtspsrc_handle_request (GstRTSPSrc * src, RTSPMessage * request)
{
RTSPMessage response = { 0 };
RTSPResult res;
res = rtsp_message_init_response (&response, RTSP_STS_OK, "OK", request);
if (res < 0)
goto send_error;
if (src->debug)
rtsp_message_dump (&response);
if ((res = rtsp_connection_send (src->connection, &response)) < 0)
goto send_error;
return RTSP_OK;
/* ERRORS */
send_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
g_free (str);
return res;
}
}
/**
* gst_rtspsrc_send:
* @src: the rtsp source
* @request: must point to a valid request
* @response: must point to an empty #RTSPMessage
*
* send @request and retrieve the response in @response. optionally @code can be
* non-NULL in which case it will contain the status code of the response.
*
* If This function returns TRUE, @response will contain a valid response
* message that should be cleaned with rtsp_message_unset() after usage.
*
* If @code is NULL, this function will return FALSE (with an invalid @response
* message) if the response code was not 200 (OK).
*
* Returns: TRUE if the processing was successful.
*/
gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPResult res;
RTSPStatusCode thecode;
gchar *content_base = NULL;
if (src->extension && src->extension->before_send)
src->extension->before_send (src->extension, request);
if (src->debug)
rtsp_message_dump (request);
if ((res = rtsp_connection_send (src->connection, request)) < 0)
goto send_error;
next:
if ((res = rtsp_connection_receive (src->connection, response)) < 0)
goto receive_error;
if (src->debug)
rtsp_message_dump (response);
switch (response->type) {
case RTSP_MESSAGE_REQUEST:
/* FIXME, handle server request, reply with OK, for now */
if ((res = gst_rtspsrc_handle_request (src, response)) < 0)
goto handle_request_failed;
goto next;
case RTSP_MESSAGE_RESPONSE:
/* ok, a response is good */
break;
default:
case RTSP_MESSAGE_DATA:
/* get next response */
goto next;
}
thecode = response->type_data.response.code;
/* if the caller wanted the result code, we store it. Else we check if it's
* OK. */
if (code)
*code = thecode;
else if (thecode != RTSP_STS_OK)
goto error_response;
/* store new content base if any */
rtsp_message_get_header (response, RTSP_HDR_CONTENT_BASE, &content_base);
if (content_base) {
g_free (src->content_base);
src->content_base = g_strdup (content_base);
}
if (src->extension && src->extension->after_send)
src->extension->after_send (src->extension, request, response);
return TRUE;
/* ERRORS */
send_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
g_free (str);
return FALSE;
}
receive_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
return FALSE;
}
handle_request_failed:
{
/* ERROR was posted */
return FALSE;
}
error_response:
{
switch (response->type_data.response.code) {
case RTSP_STS_NOT_FOUND:
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
response->type_data.response.reason));
break;
default:
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Got error response: %d (%s).", response->type_data.response.code,
response->type_data.response.reason));
break;
}
/* we return FALSE so we should unset the response ourselves */
rtsp_message_unset (response);
return FALSE;
}
}
/* parse the response and collect all the supported methods. We need this
* information so that we don't try to send an unsupported request to the
* server.
*/
static gboolean
gst_rtspsrc_parse_methods (GstRTSPSrc * src, RTSPMessage * response)
{
gchar *respoptions = NULL;
gchar **options;
gint i;
/* clear supported methods */
src->methods = 0;
/* Try Allow Header first */
rtsp_message_get_header (response, RTSP_HDR_ALLOW, &respoptions);
if (!respoptions)
/* Then maybe Public Header... */
rtsp_message_get_header (response, RTSP_HDR_PUBLIC, &respoptions);
if (!respoptions) {
/* this field is not required, assume the server supports
* DESCRIBE, SETUP and PLAY */
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
src->methods = RTSP_DESCRIBE | RTSP_SETUP | RTSP_PLAY | RTSP_PAUSE;
goto done;
}
/* If we get here, the server gave a list of supported methods, parse
* them here. The string is like:
*
* OPTIONS, DESCRIBE, ANNOUNCE, PLAY, SETUP, ...
*/
options = g_strsplit (respoptions, ",", 0);
for (i = 0; options[i]; i++) {
gchar *stripped;
gint method;
stripped = g_strstrip (options[i]);
method = rtsp_find_method (stripped);
/* keep bitfield of supported methods */
if (method != -1)
src->methods |= method;
}
g_strfreev (options);
/* we need describe and setup */
if (!(src->methods & RTSP_DESCRIBE))
goto no_describe;
if (!(src->methods & RTSP_SETUP))
goto no_setup;
done:
return TRUE;
/* ERRORS */
no_describe:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support DESCRIBE."));
return FALSE;
}
no_setup:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Server does not support SETUP."));
return FALSE;
}
}
static RTSPResult
gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
RTSPLowerTrans protocols, gchar ** transports)
{
gchar *result;
RTSPResult res;
*transports = NULL;
if (src->extension && src->extension->get_transports)
if ((res =
src->extension->get_transports (src->extension, protocols,
transports)) < 0)
goto failed;
/* extension listed transports, use those */
if (*transports != NULL)
return RTSP_OK;
result = g_strdup ("");
if (protocols & RTSP_LOWER_TRANS_UDP) {
gchar *new;
GST_DEBUG_OBJECT (src, "adding UDP unicast");
new =
g_strconcat (result, "RTP/AVP/UDP;unicast;client_port=%%u1-%%u2", NULL);
g_free (result);
result = new;
}
if (protocols & RTSP_LOWER_TRANS_UDP_MCAST) {
gchar *new;
GST_DEBUG_OBJECT (src, "adding UDP multicast");
/* we don't have to allocate any UDP ports yet, if the selected transport
* turns out to be multicast we can create them and join the multicast
* group indicated in the transport reply */
new = g_strconcat (result, result[0] ? "," : "",
"RTP/AVP/UDP;multicast", NULL);
g_free (result);
result = new;
}
if (protocols & RTSP_LOWER_TRANS_TCP) {
gchar *new;
GST_DEBUG_OBJECT (src, "adding TCP");
new = g_strconcat (result, result[0] ? "," : "",
"RTP/AVP/TCP;unicast;interleaved=%%i1-%%i2", NULL);
g_free (result);
result = new;
}
*transports = result;
return RTSP_OK;
/* ERRORS */
failed:
{
return res;
}
}
static RTSPResult
gst_rtspsrc_configure_transports (GstRTSPStream * stream, gchar ** transports)
{
GstRTSPSrc *src;
gint nr_udp, nr_int;
gchar *next, *p;
gint rtpport = 0, rtcpport = 0;
GString *str;
src = stream->parent;
/* find number of placeholders first */
if (strstr (*transports, "%%i2"))
nr_int = 2;
else if (strstr (*transports, "%%i1"))
nr_int = 1;
else
nr_int = 0;
if (strstr (*transports, "%%u2"))
nr_udp = 2;
else if (strstr (*transports, "%%u1"))
nr_udp = 1;
else
nr_udp = 0;
if (nr_udp == 0 && nr_int == 0)
goto done;
if (nr_udp > 0) {
if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
goto failed;
}
str = g_string_new ("");
p = *transports;
while ((next = strstr (p, "%%"))) {
g_string_append_len (str, p, next - p);
if (next[2] == 'u') {
if (next[3] == '1')
g_string_append_printf (str, "%d", rtpport);
else if (next[3] == '2')
g_string_append_printf (str, "%d", rtcpport);
}
if (next[2] == 'i') {
if (next[3] == '1')
g_string_append_printf (str, "%d", src->free_channel);
else if (next[3] == '2')
g_string_append_printf (str, "%d", src->free_channel + 1);
}
p = next + 4;
}
g_free (*transports);
*transports = g_string_free (str, FALSE);
done:
return RTSP_OK;
/* ERRORS */
failed:
{
return RTSP_ERROR;
}
}
static gboolean
gst_rtspsrc_open (GstRTSPSrc * src)
{
RTSPResult res;
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
guint8 *data;
guint size;
gint i, n_streams;
SDPMessage sdp = { 0 };
RTSPLowerTrans protocols;
GstRTSPStream *stream = NULL;
gchar *respcont = NULL;
/* reset our state */
src->free_channel = 0;
src->interleaved = FALSE;
gst_segment_init (&src->segment, GST_FORMAT_TIME);
/* can't continue without a valid url */
if (G_UNLIKELY (src->url == NULL))
goto no_url;
/* create connection */
GST_DEBUG_OBJECT (src, "creating connection (%s)...", src->location);
if ((res = rtsp_connection_create (src->url, &src->connection)) < 0)
goto could_not_create;
/* connect */
GST_DEBUG_OBJECT (src, "connecting (%s)...", src->location);
if ((res = rtsp_connection_connect (src->connection)) < 0)
goto could_not_connect;
/* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options...");
res = rtsp_message_init_request (&request, RTSP_OPTIONS, src->location);
if (res < 0)
goto create_request_failed;
/* send OPTIONS */
GST_DEBUG_OBJECT (src, "send options...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* parse OPTIONS */
if (!gst_rtspsrc_parse_methods (src, &response))
goto methods_error;
/* create DESCRIBE */
GST_DEBUG_OBJECT (src, "create describe...");
res = rtsp_message_init_request (&request, RTSP_DESCRIBE, src->location);
if (res < 0)
goto create_request_failed;
/* we only accept SDP for now */
rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
/* prepare global stream caps properties */
if (src->props)
gst_structure_remove_all_fields (src->props);
else
src->props = gst_structure_empty_new ("RTSP Properties");
/* send DESCRIBE */
GST_DEBUG_OBJECT (src, "send describe...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* check if reply is SDP */
rtsp_message_get_header (&response, RTSP_HDR_CONTENT_TYPE, &respcont);
/* could not be set but since the request returned OK, we assume it
* was SDP, else check it. */
if (respcont) {
if (!g_ascii_strcasecmp (respcont, "application/sdp") == 0)
goto wrong_content_type;
}
/* get message body and parse as SDP */
rtsp_message_get_body (&response, &data, &size);
GST_DEBUG_OBJECT (src, "parse SDP...");
sdp_message_init (&sdp);
sdp_message_parse_buffer (data, size, &sdp);
if (src->debug)
sdp_message_dump (&sdp);
if (src->extension && src->extension->parse_sdp)
src->extension->parse_sdp (src->extension, &sdp);
/* we initially allow all configured lower transports. based on the URL
* transports and the replies from the server we narrow them down. */
protocols = src->url->transports & src->cur_protocols;
/* setup streams */
n_streams = sdp_message_medias_len (&sdp);
for (i = 0; i < n_streams; i++) {
gchar *transports;
/* create stream from the media, can never return NULL */
stream = gst_rtspsrc_create_stream (src, &sdp, i);
/* merge/overwrite global caps */
if (stream->caps) {
guint j, num;
GstStructure *s;
s = gst_caps_get_structure (stream->caps, 0);
num = gst_structure_n_fields (src->props);
for (j = 0; j < num; j++) {
const gchar *name;
const GValue *val;
name = gst_structure_nth_field_name (src->props, j);
val = gst_structure_get_value (src->props, name);
gst_structure_set_value (s, name, val);
GST_DEBUG_OBJECT (src, "copied %s", name);
}
}
/* skip setup if we have no URL for it */
if (stream->setup_url == NULL)
continue;
GST_DEBUG_OBJECT (src, "doing setup of stream %d with %s", i,
stream->setup_url);
/* create SETUP request */
res = rtsp_message_init_request (&request, RTSP_SETUP, stream->setup_url);
if (res < 0)
goto create_request_failed;
/* create a string with all the transports */
res = gst_rtspsrc_create_transports_string (src, protocols, &transports);
if (res < 0)
goto setup_transport_failed;
/* replace placeholders with real values */
res = gst_rtspsrc_configure_transports (stream, &transports);
if (res < 0)
goto setup_transport_failed;
/* select transport, copy is made when adding to header */
rtsp_message_add_header (&request, RTSP_HDR_TRANSPORT, transports);
g_free (transports);
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* parse response transport */
{
gchar *resptrans = NULL;
RTSPTransport transport = { 0 };
rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
if (!resptrans)
goto no_transport;
/* parse transport */
if (rtsp_transport_parse (resptrans, &transport) != RTSP_OK)
continue;
/* update allowed transports for other streams. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
switch (transport.lower_transport) {
case RTSP_LOWER_TRANS_TCP:
GST_DEBUG_OBJECT (src, "stream %d as TCP interleaved", i);
protocols = RTSP_LOWER_TRANS_TCP;
src->interleaved = TRUE;
/* update free channels */
src->free_channel =
MAX (transport.interleaved.min, src->free_channel);
src->free_channel =
MAX (transport.interleaved.max, src->free_channel);
src->free_channel++;
break;
case RTSP_LOWER_TRANS_UDP_MCAST:
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UDP multicast", i);
protocols = RTSP_LOWER_TRANS_UDP_MCAST;
break;
case RTSP_LOWER_TRANS_UDP:
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UDP unicast", i);
protocols = RTSP_LOWER_TRANS_UDP;
break;
default:
GST_DEBUG_OBJECT (src, "stream %d unknown transport %d", i,
transport.lower_transport);
break;
}
if (!stream->container || !src->interleaved) {
/* now configure the stream with the transport */
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream %d transport, skipping stream", i);
}
}
/* clean up our transport struct */
rtsp_transport_init (&transport);
}
}
if (src->extension && src->extension->stream_select)
src->extension->stream_select (src->extension);
/* if we got here all was configured. We have dynamic pads so we notify that
* we are done */
gst_element_no_more_pads (GST_ELEMENT_CAST (src));
/* clean up any messages */
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return TRUE;
/* ERRORS */
no_url:
{
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
("No valid RTSP URL was provided"));
goto cleanup_error;
}
could_not_create:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not create connection. (%s)", str));
g_free (str);
goto cleanup_error;
}
could_not_connect:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
("Could not connect to server. (%s)", str));
g_free (str);
goto cleanup_error;
}
create_request_failed:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request. (%s)", str));
g_free (str);
goto cleanup_error;
}
send_error:
{
gchar *str = rtsp_strresult (res);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message. (%s)", str));
g_free (str);
goto cleanup_error;
}
methods_error:
{
/* error was posted */
goto cleanup_error;
}
wrong_content_type:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server does not support SDP, got %s.", respcont));
goto cleanup_error;
}
setup_transport_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Could not setup transport."));
goto cleanup_error;
}
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
("Server did not select transport."));
goto cleanup_error;
}
cleanup_error:
{
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return FALSE;
}
}
static gboolean
gst_rtspsrc_close (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
GST_DEBUG_OBJECT (src, "TEARDOWN...");
gst_rtspsrc_loop_send_cmd (src, CMD_STOP);
/* stop task if any */
if (src->task) {
gst_task_stop (src->task);
/* make sure it is not running */
g_static_rec_mutex_lock (src->stream_rec_lock);
g_static_rec_mutex_unlock (src->stream_rec_lock);
/* no wait for the task to finish */
gst_task_join (src->task);
/* and free the task */
gst_object_unref (GST_OBJECT (src->task));
src->task = NULL;
}
if (src->methods & RTSP_PLAY) {
/* do TEARDOWN */
res = rtsp_message_init_request (&request, RTSP_TEARDOWN, src->location);
if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* FIXME, parse result? */
rtsp_message_unset (&request);
rtsp_message_unset (&response);
}
/* close connection */
GST_DEBUG_OBJECT (src, "closing connection...");
if ((res = rtsp_connection_close (src->connection)) < 0)
goto close_failed;
return TRUE;
/* ERRORS */
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request."));
return FALSE;
}
send_error:
{
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message."));
return FALSE;
}
close_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, (NULL), ("Close failed."));
return FALSE;
}
}
/* RTP-Info is of the format:
*
* url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
*/
static gboolean
gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
{
gchar **infos;
gint i;
infos = g_strsplit (rtpinfo, ",", 0);
for (i = 0; infos[i]; i++) {
/* FIXME, do something here:
* parse url, find stream for url.
* parse seq and rtptime. The seq number should be configured in the rtp
* depayloader or session manager to detect gaps. Same for the rtptime, it
* should be used to create an initial time newsegment.
*/
}
g_strfreev (infos);
return TRUE;
}
static gboolean
gst_rtspsrc_play (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
gchar *rtpinfo;
if (!(src->methods & RTSP_PLAY))
return TRUE;
GST_DEBUG_OBJECT (src, "PLAY...");
/* do play */
res = rtsp_message_init_request (&request, RTSP_PLAY, src->location);
if (res < 0)
goto create_request_failed;
rtsp_message_add_header (&request, RTSP_HDR_RANGE, "npt=0-");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
rtsp_message_unset (&request);
/* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
* for the RTP packets. If this is not present, we assume all starts from 0...
* FIXME, this is info for the RTP session manager ideally. */
rtsp_message_get_header (&response, RTSP_HDR_RTP_INFO, &rtpinfo);
if (rtpinfo)
gst_rtspsrc_parse_rtpinfo (src, rtpinfo);
rtsp_message_unset (&response);
/* for interleaved transport, we receive the data on the RTSP connection
* instead of UDP. We start a task to select and read from that connection.
* For UDP we start the task as well to look for server info and UDP timeouts. */
if (src->task == NULL) {
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
gst_task_set_lock (src->task, src->stream_rec_lock);
}
src->running = TRUE;
gst_rtspsrc_loop_send_cmd (src, CMD_WAIT);
gst_task_start (src->task);
return TRUE;
/* ERRORS */
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request."));
return FALSE;
}
send_error:
{
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message."));
return FALSE;
}
}
static gboolean
gst_rtspsrc_pause (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
if (!(src->methods & RTSP_PAUSE))
return TRUE;
GST_DEBUG_OBJECT (src, "PAUSE...");
/* do pause */
res = rtsp_message_init_request (&request, RTSP_PAUSE, src->location);
if (res < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return TRUE;
/* ERRORS */
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
("Could not create request."));
return FALSE;
}
send_error:
{
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not send message."));
return FALSE;
}
}
static void
gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
{
switch (GST_MESSAGE_TYPE (message)) {
case GST_MESSAGE_ELEMENT:
{
GstRTSPSrc *rtspsrc;
const GstStructure *s = gst_message_get_structure (message);
rtspsrc = GST_RTSPSRC (bin);
if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
GST_DEBUG_OBJECT (bin, "timeout on UDP port");
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT);
return;
}
}
/* Fallthrough */
default:
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
break;
}
}
static GstStateChangeReturn
gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
{
GstRTSPSrc *rtspsrc;
GstStateChangeReturn ret;
rtspsrc = GST_RTSPSRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtspsrc->cur_protocols = rtspsrc->protocols;
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
rtsp_connection_flush (rtspsrc->connection, FALSE);
/* copy configuerd protocols */
gst_rtspsrc_play (rtspsrc);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
rtsp_connection_flush (rtspsrc->connection, TRUE);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
gst_rtspsrc_pause (rtspsrc);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_rtspsrc_close (rtspsrc);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
done:
return ret;
open_failed:
{
return GST_STATE_CHANGE_FAILURE;
}
}
/*** GSTURIHANDLER INTERFACE *************************************************/
static GstURIType
gst_rtspsrc_uri_get_type (void)
{
return GST_URI_SRC;
}
static gchar **
gst_rtspsrc_uri_get_protocols (void)
{
static gchar *protocols[] = { "rtsp", "rtspu", "rtspt", NULL };
return protocols;
}
static const gchar *
gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
{
GstRTSPSrc *src = GST_RTSPSRC (handler);
/* should not dup */
return src->location;
}
static gboolean
gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri)
{
GstRTSPSrc *src;
RTSPResult res;
RTSPUrl *newurl;
src = GST_RTSPSRC (handler);
/* same URI, we're fine */
if (src->location && uri && !strcmp (uri, src->location))
goto was_ok;
/* try to parse */
if ((res = rtsp_url_parse (uri, &newurl)) < 0)
goto parse_error;
/* if worked, free previous and store new url object along with the original
* location. */
rtsp_url_free (src->url);
src->url = newurl;
g_free (src->location);
src->location = g_strdup (uri);
if (!g_str_has_prefix (src->location, "rtsp://"))
memmove (src->location + 4, src->location + 5, strlen (src->location) - 4);
GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
return TRUE;
/* Special cases */
was_ok:
{
GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
return TRUE;
}
parse_error:
{
GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
GST_STR_NULL (uri), res);
return FALSE;
}
}
static void
gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtspsrc_uri_get_type;
iface->get_protocols = gst_rtspsrc_uri_get_protocols;
iface->get_uri = gst_rtspsrc_uri_get_uri;
iface->set_uri = gst_rtspsrc_uri_set_uri;
}
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