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gstreamer/gst/rtsp/gstrtspsrc.c
Wim Taymans 001a51dba2 gst/rtsp/gstrtspsrc.c: Strip spaces for key/value pairs. 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_send):
Strip spaces for key/value pairs.
2005-09-21 19:41:45 +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"
/* elementfactory information */
static GstElementDetails gst_rtspsrc_details =
GST_ELEMENT_DETAILS ("RTSP packet receiver",
"Source/Network",
"Receive data over the network via RTSP",
"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
enum
{
PROP_0,
PROP_LOCATION,
PROP_PROTOCOLS,
PROP_DEBUG,
/* 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
};
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 (G_OBJECT_CLASS (klass), 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 (G_OBJECT_CLASS (klass), 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 (G_OBJECT_CLASS (klass), PROP_DEBUG,
g_param_spec_boolean ("debug", "Debug",
"Dump request and response messages to stdout",
DEFAULT_DEBUG, 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;
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;
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;
src = stream->parent;
if (!(stream->rtpsrc =
gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL)))
goto no_udp_rtp_protocol;
/* we manage this element */
gst_rtspsrc_add_element (src, stream->rtpsrc);
ret = gst_element_set_state (stream->rtpsrc, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtp_failure;
if (!(stream->rtcpsrc =
gst_element_make_from_uri (GST_URI_SRC, "udp://0.0.0.0:0", NULL)))
goto no_udp_rtcp_protocol;
/* we manage this element */
gst_rtspsrc_add_element (src, stream->rtcpsrc);
ret = gst_element_set_state (stream->rtcpsrc, GST_STATE_PAUSED);
if (ret == GST_STATE_CHANGE_FAILURE)
goto start_rtcp_failure;
caps = gst_rtspsrc_media_to_caps (media);
g_object_set (G_OBJECT (stream->rtpsrc), "caps", caps, NULL);
g_object_get (G_OBJECT (stream->rtpsrc), "port", rtpport, NULL);
g_object_get (G_OBJECT (stream->rtcpsrc), "port", rtcpport, NULL);
return TRUE;
/* ERRORS, FIXME, cleanup */
no_udp_rtp_protocol:
{
GST_DEBUG ("could not get UDP source for rtp");
return FALSE;
}
no_udp_rtcp_protocol:
{
GST_DEBUG ("could not get UDP source for rtcp");
return FALSE;
}
start_rtp_failure:
{
GST_DEBUG ("could not start UDP source for rtp");
return FALSE;
}
start_rtcp_failure:
{
GST_DEBUG ("could not start UDP source for 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 (GST_OBJECT (pad));
pad = gst_element_get_pad (stream->rtcpsrc, "src");
gst_pad_link (pad, stream->rtpdecrtcp);
gst_object_unref (GST_OBJECT (pad));
}
pad = gst_element_get_pad (stream->rtpdec, "srcrtp");
name = g_strdup_printf ("rtp_stream%d", stream->id);
gst_element_add_pad (GST_ELEMENT (src), gst_ghost_pad_new (name, pad));
g_free (name);
gst_object_unref (GST_OBJECT (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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