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gstreamer/gst/rtsp/gstrtspsrc.c
Wim Taymans d465618d5b gst/rtsp/README: Updated README. 
Original commit message from CVS:
* gst/rtsp/README:
Updated README.

* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
(gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp):
* gst/rtsp/gstrtspsrc.h:
Make sure the RTP port is an even port an try to allocate
another if not.
Added retry property to control max retries for port allocation.
Make sure RTCP port is RTP port+1.
Cleanup when port allocation fails.
Fixes #319183.
2006-02-16 10:42:25 +00:00

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/* GStreamer
* Copyright (C) <2005> Wim Taymans <wim@fluendo.com>
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <unistd.h>
#include <string.h>
#include "gstrtspsrc.h"
#include "sdp.h"
GST_DEBUG_CATEGORY (rtspsrc_debug);
#define GST_CAT_DEFAULT (rtspsrc_debug)
/* elementfactory information */
static 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>");
static GstStaticPadTemplate rtptemplate =
GST_STATIC_PAD_TEMPLATE ("rtp_stream%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS_ANY);
static GstStaticPadTemplate rtcptemplate =
GST_STATIC_PAD_TEMPLATE ("rtcp_stream%d",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS_ANY);
enum
{
/* FILL ME */
LAST_SIGNAL
};
#define DEFAULT_LOCATION NULL
#define DEFAULT_PROTOCOLS GST_RTSP_PROTO_UDP_UNICAST | GST_RTSP_PROTO_UDP_MULTICAST | GST_RTSP_PROTO_TCP
#define DEFAULT_DEBUG FALSE
#define DEFAULT_RETRY 20
enum
{
PROP_0,
PROP_LOCATION,
PROP_PROTOCOLS,
PROP_DEBUG,
PROP_RETRY,
/* FILL ME */
};
#define GST_TYPE_RTSP_PROTO (gst_rtsp_proto_get_type())
static GType
gst_rtsp_proto_get_type (void)
{
static GType rtsp_proto_type = 0;
static GFlagsValue rtsp_proto[] = {
{GST_RTSP_PROTO_UDP_UNICAST, "UDP Unicast", "UDP unicast mode"},
{GST_RTSP_PROTO_UDP_MULTICAST, "UDP Multicast", "UDP Multicast mode"},
{GST_RTSP_PROTO_TCP, "TCP", "TCP interleaved mode"},
{0, NULL, NULL},
};
if (!rtsp_proto_type) {
rtsp_proto_type = g_flags_register_static ("GstRTSPProto", rtsp_proto);
}
return rtsp_proto_type;
}
static void gst_rtspsrc_base_init (gpointer g_class);
static void gst_rtspsrc_class_init (GstRTSPSrc * klass);
static void gst_rtspsrc_init (GstRTSPSrc * rtspsrc);
static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
GstStateChange transition);
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 void gst_rtspsrc_loop (GstRTSPSrc * src);
static GstElementClass *parent_class = NULL;
/*static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_rtspsrc_get_type (void)
{
static GType rtspsrc_type = 0;
if (!rtspsrc_type) {
static const GTypeInfo rtspsrc_info = {
sizeof (GstRTSPSrcClass),
gst_rtspsrc_base_init,
NULL,
(GClassInitFunc) gst_rtspsrc_class_init,
NULL,
NULL,
sizeof (GstRTSPSrc),
0,
(GInstanceInitFunc) gst_rtspsrc_init,
NULL
};
static const GInterfaceInfo urihandler_info = {
gst_rtspsrc_uri_handler_init,
NULL,
NULL
};
GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
rtspsrc_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstRTSPSrc", &rtspsrc_info,
0);
g_type_add_interface_static (rtspsrc_type, GST_TYPE_URI_HANDLER,
&urihandler_info);
}
return rtspsrc_type;
}
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_add_pad_template (element_class,
gst_static_pad_template_get (&rtcptemplate));
gst_element_class_set_details (element_class, &gst_rtspsrc_details);
}
static void
gst_rtspsrc_class_init (GstRTSPSrc * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
gobject_class->set_property = gst_rtspsrc_set_property;
gobject_class->get_property = gst_rtspsrc_get_property;
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_PARAM_CONSTRUCT));
g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
g_param_spec_flags ("protocols", "Protocols", "Allowed protocols",
GST_TYPE_RTSP_PROTO, 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));
gstelement_class->change_state = gst_rtspsrc_change_state;
}
static void
gst_rtspsrc_init (GstRTSPSrc * src)
{
}
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:
g_free (rtspsrc->location);
rtspsrc->location = g_value_dup_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;
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;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstRTSPStream *
gst_rtspsrc_create_stream (GstRTSPSrc * src)
{
GstRTSPStream *s;
s = g_new0 (GstRTSPStream, 1);
s->parent = src;
s->id = src->numstreams++;
src->streams = g_list_append (src->streams, s);
return s;
}
static gboolean
gst_rtspsrc_add_element (GstRTSPSrc * src, GstElement * element)
{
gst_object_set_parent (GST_OBJECT (element), GST_OBJECT (src));
return TRUE;
}
static GstStateChangeReturn
gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
{
GstStateChangeReturn ret;
GList *streams;
ret = GST_STATE_CHANGE_SUCCESS;
/* for all streams */
for (streams = src->streams; streams; streams = g_list_next (streams)) {
GstRTSPStream *stream;
stream = (GstRTSPStream *) streams->data;
/* first our rtp session manager */
if ((ret =
gst_element_set_state (stream->rtpdec,
state)) == GST_STATE_CHANGE_FAILURE)
goto done;
/* then our sources */
if (stream->rtpsrc) {
if ((ret =
gst_element_set_state (stream->rtpsrc,
state)) == GST_STATE_CHANGE_FAILURE)
goto done;
}
if (stream->rtcpsrc) {
if ((ret =
gst_element_set_state (stream->rtcpsrc,
state)) == GST_STATE_CHANGE_FAILURE)
goto done;
}
}
done:
return ret;
}
#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++;
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)
return FALSE;
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 (SDPMedia * media)
{
GstCaps *caps;
gchar *payload;
gchar *rtpmap;
gchar *fmtp;
gint pt;
gchar *name = NULL;
gint rate = -1;
gchar *params = NULL;
GstStructure *s;
payload = sdp_media_get_format (media, 0);
if (payload == NULL) {
g_warning ("payload type not given");
return NULL;
}
pt = atoi (payload);
if (pt >= 96) {
gint payload = 0;
gboolean ret;
if ((rtpmap = sdp_media_get_attribute_val (media, "rtpmap"))) {
if ((ret =
gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate,
&params))) {
if (payload != pt) {
g_warning ("rtpmap of wrong payload type");
name = NULL;
rate = -1;
params = NULL;
}
} else {
g_warning ("error parsing rtpmap");
}
} else {
g_warning ("rtpmap type not given fot dynamic payload %d", pt);
return NULL;
}
}
caps = gst_caps_new_simple ("application/x-rtp",
"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;
PARSE_INT (p, " ", payload);
if (payload != -1 && payload == pt) {
gchar **pairs;
gint i;
pairs = g_strsplit (p, ";", 0);
for (i = 0; pairs[i]; i++) {
gchar **keyval;
keyval = g_strsplit (pairs[i], "=", 0);
if (keyval[0]) {
gchar *val, *key;
if (keyval[1])
val = g_strstrip (keyval[1]);
else
val = "1";
key = g_strstrip (keyval[0]);
gst_structure_set (s, key, G_TYPE_STRING, val, NULL);
}
g_strfreev (keyval);
}
g_strfreev (pairs);
}
}
return caps;
}
static gboolean
gst_rtspsrc_stream_setup_rtp (GstRTSPStream * stream, SDPMedia * media,
gint * rtpport, gint * rtcpport)
{
GstStateChangeReturn ret;
GstRTSPSrc *src;
GstCaps *caps;
GstElement *tmp, *rtp, *rtcp;
gint tmp_rtp, tmp_rtcp;
guint count;
src = stream->parent;
tmp = NULL;
rtp = NULL;
rtcp = 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:
rtp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL);
if (rtp == NULL)
goto no_udp_rtp_protocol;
ret = gst_element_set_state (rtp, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtp_failure;
g_object_get (G_OBJECT (rtp), "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 = rtp;
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 */
rtcp = gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0", NULL);
if (rtcp == NULL)
goto no_udp_rtcp_protocol;
/* set port */
tmp_rtcp = tmp_rtp + 1;
g_object_set (G_OBJECT (rtcp), "port", tmp_rtcp, NULL);
GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
ret = gst_element_set_state (rtcp, 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 (rtp), "port", rtpport, NULL);
g_object_get (G_OBJECT (rtcp), "port", rtcpport, NULL);
/* this should not happen */
if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
goto port_error;
/* we manage these elements */
stream->rtpsrc = rtp;
gst_rtspsrc_add_element (src, stream->rtpsrc);
stream->rtcpsrc = rtcp;
gst_rtspsrc_add_element (src, stream->rtcpsrc);
caps = gst_rtspsrc_media_to_caps (media);
/* set caps */
g_object_set (G_OBJECT (stream->rtpsrc), "caps", caps, NULL);
return TRUE;
/* ERRORS */
no_udp_rtp_protocol:
{
GST_DEBUG ("could not get UDP source for RTP");
goto cleanup;
}
start_rtp_failure:
{
GST_DEBUG ("could not start UDP source for RTP");
goto cleanup;
}
no_ports:
{
GST_DEBUG ("could not allocate UDP port pair after %d retries", count);
goto cleanup;
}
no_udp_rtcp_protocol:
{
GST_DEBUG ("could not get UDP source for RTCP");
goto cleanup;
}
start_rtcp_failure:
{
GST_DEBUG ("could not start UDP source for RTCP");
goto cleanup;
}
port_error:
{
GST_DEBUG ("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 (rtp) {
gst_element_set_state (rtp, GST_STATE_NULL);
gst_object_unref (rtp);
}
if (rtcp) {
gst_element_set_state (rtcp, GST_STATE_NULL);
gst_object_unref (rtcp);
}
return FALSE;
}
}
static gboolean
gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
RTSPTransport * transport)
{
GstRTSPSrc *src;
GstPad *pad;
GstStateChangeReturn ret;
gchar *name;
src = stream->parent;
if (!(stream->rtpdec = gst_element_factory_make ("rtpdec", NULL)))
goto no_element;
/* we manage this element */
gst_rtspsrc_add_element (src, stream->rtpdec);
if ((ret =
gst_element_set_state (stream->rtpdec,
GST_STATE_PAUSED)) != GST_STATE_CHANGE_SUCCESS)
goto start_rtpdec_failure;
stream->rtpdecrtp = gst_element_get_pad (stream->rtpdec, "sinkrtp");
stream->rtpdecrtcp = gst_element_get_pad (stream->rtpdec, "sinkrtcp");
if (transport->lower_transport == RTSP_LOWER_TRANS_TCP) {
/* configure for interleaved delivery, nothing needs to be done
* here, the loop function will call the chain functions of the
* rtp session manager. */
} else {
/* configure for UDP delivery, we need to connect the udp pads to
* the rtp session plugin. */
pad = gst_element_get_pad (stream->rtpsrc, "src");
gst_pad_link (pad, stream->rtpdecrtp);
gst_object_unref (pad);
pad = gst_element_get_pad (stream->rtcpsrc, "src");
gst_pad_link (pad, stream->rtpdecrtcp);
gst_object_unref (pad);
}
pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
name = g_strdup_printf ("rtp_stream%d", stream->id);
gst_element_add_pad (GST_ELEMENT_CAST (src), gst_ghost_pad_new (name, pad));
g_free (name);
gst_object_unref (pad);
return TRUE;
no_element:
{
GST_DEBUG ("no rtpdec element found");
return FALSE;
}
start_rtpdec_failure:
{
GST_DEBUG ("could not start RTP session");
return FALSE;
}
}
static gint
find_stream (GstRTSPStream * stream, gconstpointer a)
{
gint channel = GPOINTER_TO_INT (a);
if (stream->rtpchannel == channel || stream->rtcpchannel == channel)
return 0;
return -1;
}
static void
gst_rtspsrc_loop (GstRTSPSrc * src)
{
RTSPMessage response = { 0 };
RTSPResult res;
gint channel;
GList *lstream;
GstRTSPStream *stream;
GstPad *outpad = NULL;
guint8 *data;
guint size;
do {
GST_DEBUG ("doing reveive");
if ((res = rtsp_connection_receive (src->connection, &response)) < 0)
goto receive_error;
GST_DEBUG ("got packet");
}
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);
if (!lstream)
goto unknown_stream;
stream = (GstRTSPStream *) lstream->data;
if (channel == stream->rtpchannel)
outpad = stream->rtpdecrtp;
else if (channel == stream->rtcpchannel)
outpad = stream->rtpdecrtcp;
rtsp_message_get_body (&response, &data, &size);
/* channels are not correct on some servers, do extra check */
if (data[1] >= 200 && data[1] <= 204) {
/* hmm RTCP message */
outpad = stream->rtpdecrtcp;
}
/* we have no clue what this is, just ignore then. */
if (outpad == NULL)
goto unknown_stream;
/* and chain buffer to internal element */
{
GstBuffer *buf;
buf = gst_buffer_new_and_alloc (size);
memcpy (GST_BUFFER_DATA (buf), data, size);
if (gst_pad_chain (outpad, buf) != GST_FLOW_OK)
goto need_pause;
}
unknown_stream:
return;
receive_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not receive message."), (NULL));
/*
gst_pad_push_event (src->srcpad, gst_event_new (GST_EVENT_EOS));
*/
goto need_pause;
}
need_pause:
{
gst_task_pause (src->task);
return;
}
}
static gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPResult res;
if (src->debug) {
rtsp_message_dump (request);
}
if ((res = rtsp_connection_send (src->connection, request)) < 0)
goto send_error;
if ((res = rtsp_connection_receive (src->connection, response)) < 0)
goto receive_error;
if (code) {
*code = response->type_data.response.code;
}
if (src->debug) {
rtsp_message_dump (response);
}
if (response->type_data.response.code != RTSP_STS_OK)
goto error_response;
return TRUE;
send_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
}
receive_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ,
("Could not receive message."), (NULL));
return FALSE;
}
error_response:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, ("Got error response: %d (%s).",
response->type_data.response.code,
response->type_data.response.reason), (NULL));
return FALSE;
}
}
static gboolean
gst_rtspsrc_open (GstRTSPSrc * src)
{
RTSPUrl *url;
RTSPResult res;
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
guint8 *data;
guint size;
SDPMessage sdp = { 0 };
GstRTSPProto protocols;
/* parse url */
GST_DEBUG ("parsing url...");
if ((res = rtsp_url_parse (src->location, &url)) < 0)
goto invalid_url;
/* open connection */
GST_DEBUG ("opening connection...");
if ((res = rtsp_connection_open (url, &src->connection)) < 0)
goto could_not_open;
/* create OPTIONS */
GST_DEBUG ("create options...");
if ((res =
rtsp_message_init_request (RTSP_OPTIONS, src->location,
&request)) < 0)
goto create_request_failed;
/* send OPTIONS */
GST_DEBUG ("send options...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
{
gchar *respoptions = NULL;
gchar **options;
gint i;
/* 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) {
goto no_options;
}
}
/* parse options */
options = g_strsplit (respoptions, ",", 0);
i = 0;
while (options[i]) {
gchar *stripped;
gint method;
stripped = g_strdup (options[i]);
stripped = g_strstrip (stripped);
method = rtsp_find_method (stripped);
g_free (stripped);
/* keep bitfield of supported methods */
if (method != -1)
src->options |= method;
i++;
}
g_strfreev (options);
/* we need describe and setup */
if (!(src->options & RTSP_DESCRIBE))
goto no_describe;
if (!(src->options & RTSP_SETUP))
goto no_setup;
}
/* create DESCRIBE */
GST_DEBUG ("create describe...");
if ((res =
rtsp_message_init_request (RTSP_DESCRIBE, src->location,
&request)) < 0)
goto create_request_failed;
/* we accept SDP for now */
rtsp_message_add_header (&request, RTSP_HDR_ACCEPT, "application/sdp");
/* send DESCRIBE */
GST_DEBUG ("send describe...");
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
/* check if reply is SDP */
{
gchar *respcont = NULL;
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;
}
}
/* parse SDP */
rtsp_message_get_body (&response, &data, &size);
GST_DEBUG ("parse sdp...");
sdp_message_init (&sdp);
sdp_message_parse_buffer (data, size, &sdp);
if (src->debug)
sdp_message_dump (&sdp);
/* we allow all configured protocols */
protocols = src->protocols;
/* setup streams */
{
gint i;
for (i = 0; i < sdp_message_medias_len (&sdp); i++) {
SDPMedia *media;
gchar *setup_url;
gchar *control_url;
gchar *transports;
GstRTSPStream *stream;
media = sdp_message_get_media (&sdp, i);
stream = gst_rtspsrc_create_stream (src);
GST_DEBUG ("setup media %d", i);
control_url = sdp_media_get_attribute_val (media, "control");
if (control_url == NULL) {
GST_DEBUG ("no control url found, skipping stream");
continue;
}
/* check absolute/relative URL */
/* FIXME, what if the control_url starts with a '/' or a non rtsp: protocol? */
if (g_str_has_prefix (control_url, "rtsp://")) {
setup_url = g_strdup (control_url);
} else {
setup_url = g_strdup_printf ("%s/%s", src->location, control_url);
}
GST_DEBUG ("setup %s", setup_url);
/* create SETUP request */
if ((res =
rtsp_message_init_request (RTSP_SETUP, setup_url,
&request)) < 0) {
g_free (setup_url);
goto create_request_failed;
}
g_free (setup_url);
transports = g_strdup ("");
if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
gchar *new;
gint rtpport, rtcpport;
gchar *trxparams;
/* allocate two udp ports */
if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
goto setup_rtp_failed;
trxparams = g_strdup_printf ("client_port=%d-%d", rtpport, rtcpport);
new = g_strconcat (transports, "RTP/AVP/UDP;unicast;", trxparams, NULL);
g_free (trxparams);
g_free (transports);
transports = new;
}
if (protocols & GST_RTSP_PROTO_UDP_MULTICAST) {
gchar *new;
new =
g_strconcat (transports, transports[0] ? "," : "",
"RTP/AVP/UDP;multicast", NULL);
g_free (transports);
transports = new;
}
if (protocols & GST_RTSP_PROTO_TCP) {
gchar *new;
new =
g_strconcat (transports, transports[0] ? "," : "", "RTP/AVP/TCP",
NULL);
g_free (transports);
transports = new;
}
/* 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 */
rtsp_transport_parse (resptrans, &transport);
/* update allowed transports for other streams */
if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
protocols = GST_RTSP_PROTO_TCP;
src->interleaved = TRUE;
} else {
if (transport.multicast) {
/* disable unicast */
protocols = GST_RTSP_PROTO_UDP_MULTICAST;
} else {
/* disable multicast */
protocols = GST_RTSP_PROTO_UDP_UNICAST;
}
}
/* now configure the stream with the transport */
if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
GST_DEBUG ("could not configure stream transport, skipping stream");
}
/* clean up our transport struct */
rtsp_transport_init (&transport);
}
}
}
return TRUE;
/* ERRORS */
invalid_url:
{
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
("Not a valid RTSP url."), (NULL));
return FALSE;
}
could_not_open:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE,
("Could not open connection."), (NULL));
return FALSE;
}
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT,
("Could not create request."), (NULL));
return FALSE;
}
send_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
}
no_options:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Invalid OPTIONS response."), (NULL));
return FALSE;
}
no_describe:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server does not support DESCRIBE."), (NULL));
return FALSE;
}
no_setup:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server does not support SETUP."), (NULL));
return FALSE;
}
wrong_content_type:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server does not support SDP."), (NULL));
return FALSE;
}
setup_rtp_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, ("Could not setup rtp."), (NULL));
return FALSE;
}
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Server did not select transport."), (NULL));
return FALSE;
}
}
static gboolean
gst_rtspsrc_close (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
GST_DEBUG ("TEARDOWN...");
/* stop task if any */
if (src->task) {
gst_task_stop (src->task);
gst_object_unref (GST_OBJECT (src->task));
src->task = NULL;
}
if (src->options & RTSP_PLAY) {
/* do TEARDOWN */
if ((res =
rtsp_message_init_request (RTSP_TEARDOWN, src->location,
&request)) < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
}
/* close connection */
GST_DEBUG ("closing connection...");
if ((res = rtsp_connection_close (src->connection)) < 0)
goto close_failed;
return TRUE;
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT,
("Could not create request."), (NULL));
return FALSE;
}
send_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
}
close_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, CLOSE, ("Close failed."), (NULL));
return FALSE;
}
}
static gboolean
gst_rtspsrc_play (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
if (!(src->options & RTSP_PLAY))
return TRUE;
GST_DEBUG ("PLAY...");
/* do play */
if ((res =
rtsp_message_init_request (RTSP_PLAY, src->location, &request)) < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
if (src->interleaved) {
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
gst_task_start (src->task);
}
return TRUE;
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT,
("Could not create request."), (NULL));
return FALSE;
}
send_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
}
}
static gboolean
gst_rtspsrc_pause (GstRTSPSrc * src)
{
RTSPMessage request = { 0 };
RTSPMessage response = { 0 };
RTSPResult res;
if (!(src->options & RTSP_PAUSE))
return TRUE;
GST_DEBUG ("PAUSE...");
/* do pause */
if ((res =
rtsp_message_init_request (RTSP_PAUSE, src->location, &request)) < 0)
goto create_request_failed;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
return TRUE;
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT,
("Could not create request."), (NULL));
return FALSE;
}
send_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
}
}
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->interleaved = FALSE;
rtspsrc->options = 0;
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
gst_rtspsrc_play (rtspsrc);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
goto done;
ret = gst_rtspsrc_set_state (rtspsrc, GST_STATE_PENDING (rtspsrc));
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 guint
gst_rtspsrc_uri_get_type (void)
{
return GST_URI_SRC;
}
static gchar **
gst_rtspsrc_uri_get_protocols (void)
{
static gchar *protocols[] = { "rtsp", NULL };
return protocols;
}
static const gchar *
gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
{
GstRTSPSrc *src = GST_RTSPSRC (handler);
return g_strdup (src->location);
}
static gboolean
gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri)
{
GstRTSPSrc *src = GST_RTSPSRC (handler);
g_free (src->location);
src->location = g_strdup (uri);
return TRUE;
}
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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