FaSTTUBe/GST-Tensordecoder-ov_ep
37
0
Fork 0
Code Issues Pull Requests 1 Actions Packages Projects Releases Wiki Activity

gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation.

Browse Source 
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Small cleanups, added documentation.
Try to clean up the requests and responses.
Refactor parsing the supported methods.
* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
(rtsp_connection_create), (rtsp_connection_send),
(parse_response_status), (parse_request_line),
(rtsp_connection_receive), (rtsp_connection_close),
(rtsp_connection_free):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (rtsp_transport_parse),
(rtsp_transport_free):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
(sdp_message_clean), (sdp_message_free), (sdp_media_new),
(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
Use g_return_val some more.
* gst/rtsp/rtspdefs.h:
Add more enum values to track initial states.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
(rtsp_message_init_request), (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_add_header), (rtsp_message_remove_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_take_body), (rtsp_message_get_body),
(rtsp_message_steal_body), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Reorder arguments, object goes as the first one.
Use g_return_val some more.
This commit is contained in:
Wim Taymans 2006-09-18 17:37:46 +00:00
parent ee58147b3d
commit a437e9f0ed
10 changed files with 467 additions and 306 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

41
ChangeLog
View File

@ -1,3 +1,44 @@
2006-09-18 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Small cleanups, added documentation.
Try to clean up the requests and responses.
Refactor parsing the supported methods.
* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
(rtsp_connection_create), (rtsp_connection_send),
(parse_response_status), (parse_request_line),
(rtsp_connection_receive), (rtsp_connection_close),
(rtsp_connection_free):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (rtsp_transport_parse),
(rtsp_transport_free):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
(sdp_message_clean), (sdp_message_free), (sdp_media_new),
(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
Use g_return_val some more.
* gst/rtsp/rtspdefs.h:
Add more enum values to track initial states.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
(rtsp_message_init_request), (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_add_header), (rtsp_message_remove_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_take_body), (rtsp_message_get_body),
(rtsp_message_steal_body), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Reorder arguments, object goes as the first one.
Use g_return_val some more.
2006-09-18 Wim Taymans <wim@fluendo.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),

505
gst/rtsp/gstrtspsrc.c
View File

@ -354,6 +354,10 @@ G_STMT_START { \
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)
@ -413,10 +417,9 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
GstStructure *s;
payload = sdp_media_get_format (media, 0);
if (payload == NULL) {
g_warning ("payload type not given");
return NULL;
}
if (payload == NULL)
goto no_payload;
pt = atoi (payload);
/* dynamic payloads need rtpmap */
@ -428,18 +431,18 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
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 {
g_warning ("rtpmap type not given for dynamic payload %d", pt);
return NULL;
}
} else
goto no_rtpmap;
}
caps = gst_caps_new_simple ("application/x-rtp",
@ -494,6 +497,18 @@ gst_rtspsrc_media_to_caps (SDPMedia * media)
}
}
return caps;
/* ERRORS */
no_payload:
{
g_warning ("payload type not given");
return NULL;
}
no_rtpmap:
{
g_warning ("rtpmap type not given for dynamic payload %d", pt);
return NULL;
}
}
static gboolean
@ -948,29 +963,48 @@ need_pause:
}
}
/**
* 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.
*/
static gboolean
gst_rtspsrc_send (GstRTSPSrc * src, RTSPMessage * request,
RTSPMessage * response, RTSPStatusCode * code)
{
RTSPResult res;
RTSPStatusCode thecode;
if (src->debug) {
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) {
if (src->debug)
rtsp_message_dump (response);
}
if (response->type_data.response.code != RTSP_STS_OK)
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;
return TRUE;
@ -990,9 +1024,92 @@ receive_error:
}
error_response:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, ("Got error response: %d (%s).",
response->type_data.response.code,
response->type_data.response.reason), (NULL));
switch (response->type_data.response.code) {
case RTSP_STS_NOT_FOUND:
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, ("%s",
response->type_data.response.reason), (NULL));
break;
default:
GST_ELEMENT_ERROR (src, RESOURCE, READ, ("Got error response: %d (%s).",
response->type_data.response.code,
response->type_data.response.reason), (NULL));
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 and SETUP*/
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
src->methods = RTSP_DESCRIBE | RTSP_SETUP;
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);
i = 0;
while (options[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;
i++;
}
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,
("Server does not support DESCRIBE."), (NULL));
return FALSE;
}
no_setup:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
("Server does not support SETUP."), (NULL));
return FALSE;
}
}
@ -1005,11 +1122,13 @@ gst_rtspsrc_open (GstRTSPSrc * src)
RTSPMessage response = { 0 };
guint8 *data;
guint size;
gint i, n_streams;
SDPMessage sdp = { 0 };
GstRTSPProto protocols;
GstRTSPStream *stream = NULL;
/* can't continue without a valid url */
if (src->url == NULL)
if (G_UNLIKELY (src->url == NULL))
goto no_url;
/* open connection */
@ -1019,7 +1138,7 @@ gst_rtspsrc_open (GstRTSPSrc * src)
/* create OPTIONS */
GST_DEBUG_OBJECT (src, "create options...");
res = rtsp_message_init_request (RTSP_OPTIONS, src->location, &request);
res = rtsp_message_init_request (&request, RTSP_OPTIONS, src->location);
if (res < 0)
goto create_request_failed;
@ -1028,56 +1147,13 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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) {
/* this field is not required, assume the server supports
* DESCRIBE and SETUP*/
GST_DEBUG_OBJECT (src, "could not get OPTIONS");
src->options = RTSP_DESCRIBE | RTSP_SETUP;
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);
no_options:
/* we need describe and setup */
if (!(src->options & RTSP_DESCRIBE))
goto no_describe;
if (!(src->options & RTSP_SETUP))
goto no_setup;
}
/* 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 (RTSP_DESCRIBE, src->location, &request);
res = rtsp_message_init_request (&request, RTSP_DESCRIBE, src->location);
if (res < 0)
goto create_request_failed;
@ -1117,134 +1193,138 @@ gst_rtspsrc_open (GstRTSPSrc * src)
protocols = src->protocols;
/* setup streams */
{
gint i;
n_streams = sdp_message_medias_len (&sdp);
for (i = 0; i < n_streams; i++) {
SDPMedia *media;
gchar *setup_url;
gchar *control_url;
gchar *transports;
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);
media = sdp_message_get_media (&sdp, i);
GST_DEBUG_OBJECT (src, "setup media %d", i);
control_url = sdp_media_get_attribute_val (media, "control");
if (control_url == NULL) {
GST_DEBUG_OBJECT (src, "no control url found, skipping stream %d", i);
continue;
}
stream = gst_rtspsrc_create_stream (src);
/* 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_OBJECT (src, "setup media %d", i);
control_url = sdp_media_get_attribute_val (media, "control");
if (control_url == NULL) {
GST_DEBUG_OBJECT (src, "no control url found, skipping stream");
continue;
}
GST_DEBUG_OBJECT (src, "setup %s", setup_url);
/* 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);
}
/* create SETUP request */
res = rtsp_message_init_request (&request, RTSP_SETUP, setup_url);
g_free (setup_url);
if (res < 0)
goto create_request_failed;
GST_DEBUG_OBJECT (src, "setup %s", setup_url);
stream = gst_rtspsrc_create_stream (src);
/* create SETUP request */
res = rtsp_message_init_request (RTSP_SETUP, setup_url, &request);
g_free (setup_url);
if (res < 0)
goto create_request_failed;
transports = g_strdup ("");
if (protocols & GST_RTSP_PROTO_UDP_UNICAST) {
gchar *new;
gint rtpport, rtcpport;
gchar *trxparams;
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;
/* allocate two UDP ports */
if (!gst_rtspsrc_stream_setup_rtp (stream, media, &rtpport, &rtcpport))
goto setup_rtp_failed;
GST_DEBUG_OBJECT (src, "setting up RTP ports %d-%d", rtpport, rtcpport);
GST_DEBUG_OBJECT (src, "setting up RTP ports %d-%d", rtpport, rtcpport);
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;
GST_DEBUG_OBJECT (src, "setting up MULTICAST");
/* we don't hav 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 (transports, transports[0] ? "," : "",
"RTP/AVP/UDP;multicast", NULL);
g_free (transports);
transports = new;
}
if (protocols & GST_RTSP_PROTO_TCP) {
gchar *new;
GST_DEBUG_OBJECT (src, "setting up TCP");
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);
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;
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
GST_DEBUG_OBJECT (src, "setting up MULTICAST");
/* parse response transport */
{
gchar *resptrans = NULL;
RTSPTransport transport = { 0 };
/* we don't hav 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 (transports, transports[0] ? "," : "",
"RTP/AVP/UDP;multicast", NULL);
g_free (transports);
transports = new;
}
if (protocols & GST_RTSP_PROTO_TCP) {
gchar *new;
rtsp_message_get_header (&response, RTSP_HDR_TRANSPORT, &resptrans);
if (!resptrans)
goto no_transport;
GST_DEBUG_OBJECT (src, "setting up TCP");
/* parse transport */
rtsp_transport_parse (resptrans, &transport);
new =
g_strconcat (transports, transports[0] ? "," : "", "RTP/AVP/TCP",
NULL);
g_free (transports);
transports = new;
}
/* 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 */
if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (src, "stream %d as TCP", i);
protocols = GST_RTSP_PROTO_TCP;
src->interleaved = TRUE;
/* 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. once the transport of
* one stream has been determined, we make sure that all other streams
* are configured in the same way */
if (transport.lower_transport == RTSP_LOWER_TRANS_TCP) {
GST_DEBUG_OBJECT (src, "stream %d as TCP", i);
protocols = GST_RTSP_PROTO_TCP;
src->interleaved = TRUE;
} else {
if (transport.multicast) {
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as MULTICAST", i);
protocols = GST_RTSP_PROTO_UDP_MULTICAST;
} else {
if (transport.multicast) {
/* only allow multicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as MULTICAST", i);
protocols = GST_RTSP_PROTO_UDP_MULTICAST;
} else {
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UNICAST", i);
protocols = GST_RTSP_PROTO_UDP_UNICAST;
}
/* only allow unicast for other streams */
GST_DEBUG_OBJECT (src, "stream %d as UNICAST", i);
protocols = GST_RTSP_PROTO_UDP_UNICAST;
}
/* now configure the stream with the transport */
if (!gst_rtspsrc_stream_configure_transport (stream, media, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream transport, skipping stream");
}
/* clean up our transport struct */
rtsp_transport_init (&transport);
}
/* now configure the stream with the transport */
if (!gst_rtspsrc_stream_configure_transport (stream, media, &transport)) {
GST_DEBUG_OBJECT (src,
"could not configure stream %d transport, skipping stream", i);
}
/* clean up our transport struct */
rtsp_transport_init (&transport);
}
}
/* 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 */
@ -1252,53 +1332,53 @@ no_url:
{
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
("No valid RTSP url was provided"), (NULL));
return FALSE;
goto cleanup_error;
}
could_not_open:
{
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE,
("Could not open connection."), (NULL));
return FALSE;
goto cleanup_error;
}
create_request_failed:
{
GST_ELEMENT_ERROR (src, LIBRARY, INIT,
("Could not create request."), (NULL));
return FALSE;
goto cleanup_error;
}
send_error:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
goto cleanup_error;
}
no_describe:
methods_error:
{
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;
/* error was posted */
goto cleanup_error;
}
wrong_content_type:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
("Server does not support SDP."), (NULL));
return FALSE;
goto cleanup_error;
}
setup_rtp_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, ("Could not setup rtp."), (NULL));
return FALSE;
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, ("Could not setup rtp."),
(NULL));
goto cleanup_error;
}
no_transport:
{
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
("Server did not select transport."), (NULL));
goto cleanup_error;
}
cleanup_error:
{
rtsp_message_unset (&request);
rtsp_message_unset (&response);
return FALSE;
}
}
@ -1328,14 +1408,18 @@ gst_rtspsrc_close (GstRTSPSrc * src)
src->task = NULL;
}
if (src->options & RTSP_PLAY) {
if (src->methods & RTSP_PLAY) {
/* do TEARDOWN */
res = rtsp_message_init_request (RTSP_TEARDOWN, src->location, &request);
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 */
@ -1354,6 +1438,7 @@ create_request_failed:
}
send_error:
{
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
@ -1372,13 +1457,13 @@ gst_rtspsrc_play (GstRTSPSrc * src)
RTSPMessage response = { 0 };
RTSPResult res;
if (!(src->options & RTSP_PLAY))
if (!(src->methods & RTSP_PLAY))
return TRUE;
GST_DEBUG_OBJECT (src, "PLAY...");
/* do play */
res = rtsp_message_init_request (RTSP_PLAY, src->location, &request);
res = rtsp_message_init_request (&request, RTSP_PLAY, src->location);
if (res < 0)
goto create_request_failed;
@ -1387,10 +1472,16 @@ gst_rtspsrc_play (GstRTSPSrc * src)
if (!gst_rtspsrc_send (src, &request, &response, NULL))
goto send_error;
rtsp_message_unset (&request);
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. */
if (src->interleaved) {
src->task = gst_task_create ((GstTaskFunction) gst_rtspsrc_loop, src);
gst_task_set_lock (src->task, src->stream_rec_lock);
src->running = TRUE;
gst_task_start (src->task);
}
@ -1405,6 +1496,7 @@ create_request_failed:
}
send_error:
{
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
@ -1418,18 +1510,21 @@ gst_rtspsrc_pause (GstRTSPSrc * src)
RTSPMessage response = { 0 };
RTSPResult res;
if (!(src->options & RTSP_PAUSE))
if (!(src->methods & RTSP_PAUSE))
return TRUE;
GST_DEBUG_OBJECT (src, "PAUSE...");
/* do pause */
res = rtsp_message_init_request (RTSP_PAUSE, src->location, &request);
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 */
@ -1441,6 +1536,7 @@ create_request_failed:
}
send_error:
{
rtsp_message_unset (&request);
GST_ELEMENT_ERROR (src, RESOURCE, WRITE,
("Could not send message."), (NULL));
return FALSE;
@ -1461,7 +1557,7 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
rtspsrc->interleaved = FALSE;
rtspsrc->options = 0;
gst_segment_init (&rtspsrc->segment, GST_FORMAT_TIME);
if (!gst_rtspsrc_open (rtspsrc))
goto open_failed;
break;
@ -1521,6 +1617,7 @@ gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
{
GstRTSPSrc *src = GST_RTSPSRC (handler);
/* should not dup */
return src->location;
}
@ -1560,8 +1657,8 @@ was_ok:
}
parse_error:
{
GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)", GST_STR_NULL (uri),
res);
GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
GST_STR_NULL (uri), res);
return FALSE;
}
}

4
gst/rtsp/gstrtspsrc.h
View File

@ -103,8 +103,8 @@ struct _GstRTSPSrc {
guint retry;
GstRTSPProto protocols;
/* supported options */
gint options;
/* supported methods */
gint methods;
RTSPConnection *connection;
RTSPMessage *request;

26
gst/rtsp/rtspconnection.c
View File

@ -71,8 +71,8 @@ rtsp_connection_open (RTSPUrl * url, RTSPConnection ** conn)
struct in_addr addr;
gint ret;
if (url == NULL || conn == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (url != NULL, RTSP_EINVAL);
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
if (url->protocol != RTSP_PROTO_TCP)
return RTSP_ENOTIMPL;
@ -128,6 +128,8 @@ rtsp_connection_create (gint fd, RTSPConnection ** conn)
{
RTSPConnection *newconn;
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
/* FIXME check fd, must be connected SOCK_STREAM */
newconn = g_new (RTSPConnection, 1);
@ -157,8 +159,8 @@ rtsp_connection_send (RTSPConnection * conn, RTSPMessage * message)
gint towrite;
gchar *data;
if (conn == NULL || message == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
g_return_val_if_fail (message != NULL, RTSP_EINVAL);
#ifdef G_OS_WIN32
WSADATA w;
@ -333,7 +335,7 @@ parse_response_status (gchar * buffer, RTSPMessage * msg)
while (g_ascii_isspace (*bptr))
bptr++;
rtsp_message_init_response (code, bptr, NULL, msg);
rtsp_message_init_response (msg, code, bptr, NULL);
return RTSP_OK;
@ -365,7 +367,7 @@ parse_request_line (gchar * buffer, RTSPMessage * msg)
if (strcmp (versionstr, "RTSP/1.0") != 0)
goto wrong_version;
rtsp_message_init_request (method, urlstr, msg);
rtsp_message_init_request (msg, method, urlstr);
return RTSP_OK;
@ -456,8 +458,8 @@ rtsp_connection_receive (RTSPConnection * conn, RTSPMessage * msg)
RTSPResult res;
gboolean need_body;
if (conn == NULL || msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
line = 0;
@ -490,7 +492,7 @@ rtsp_connection_receive (RTSPConnection * conn, RTSPMessage * msg)
goto error;
/* now we create a data message */
rtsp_message_init_data ((gint) c, msg);
rtsp_message_init_data (msg, (gint) c);
/* next two bytes are the length of the data */
ret = read (conn->fd, &size, 2);
@ -590,8 +592,7 @@ rtsp_connection_close (RTSPConnection * conn)
{
gint res;
if (conn == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
res = CLOSE_SOCKET (conn->fd);
#ifdef G_OS_WIN32
@ -612,8 +613,7 @@ sys_error:
RTSPResult
rtsp_connection_free (RTSPConnection * conn)
{
if (conn == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (conn != NULL, RTSP_EINVAL);
#ifdef G_OS_WIN32
WSACleanup ();

3
gst/rtsp/rtspdefs.h
View File

@ -50,6 +50,7 @@ typedef enum {
} RTSPFamily;
typedef enum {
RTSP_STATE_INVALID,
RTSP_STATE_INIT,
RTSP_STATE_READY,
RTSP_STATE_PLAYING,
@ -57,6 +58,7 @@ typedef enum {
} RTSPState;
typedef enum {
RTSP_INVALID = 0,
RTSP_DESCRIBE = (1 << 0),
RTSP_ANNOUNCE = (1 << 1),
RTSP_GET_PARAMETER = (1 << 2),
@ -120,6 +122,7 @@ typedef enum {
} RTSPHeaderField;
typedef enum {
RTSP_STS_INVALID = 0,
RTSP_STS_CONTINUE = 100,
RTSP_STS_OK = 200,
RTSP_STS_CREATED = 201,

132
gst/rtsp/rtspmessage.c
View File

@ -20,84 +20,69 @@
#include "rtspmessage.h"
RTSPResult
rtsp_message_new_request (RTSPMethod method, gchar * uri, RTSPMessage ** msg)
rtsp_message_new_request (RTSPMessage ** msg, RTSPMethod method, gchar * uri)
{
RTSPMessage *newmsg;
if (msg == NULL || uri == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (uri != NULL, RTSP_EINVAL);
newmsg = g_new0 (RTSPMessage, 1);
*msg = newmsg;
return rtsp_message_init_request (method, uri, newmsg);
return rtsp_message_init_request (newmsg, method, uri);
}
RTSPResult
rtsp_message_init_request (RTSPMethod method, gchar * uri, RTSPMessage * msg)
rtsp_message_init_request (RTSPMessage * msg, RTSPMethod method, gchar * uri)
{
if (msg == NULL || uri == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (uri != NULL, RTSP_EINVAL);
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
rtsp_message_unset (msg);
msg->type = RTSP_MESSAGE_REQUEST;
msg->type_data.request.method = method;
g_free (msg->type_data.request.uri);
msg->type_data.request.uri = g_strdup (uri);
if (msg->hdr_fields != NULL)
g_hash_table_destroy (msg->hdr_fields);
msg->hdr_fields =
g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL, g_free);
if (msg->body) {
g_free (msg->body);
msg->body = NULL;
}
msg->body_size = 0;
return RTSP_OK;
}
RTSPResult
rtsp_message_new_response (RTSPStatusCode code, gchar * reason,
RTSPMessage * request, RTSPMessage ** msg)
rtsp_message_new_response (RTSPMessage ** msg, RTSPStatusCode code,
gchar * reason, RTSPMessage * request)
{
RTSPMessage *newmsg;
if (msg == NULL || reason == NULL || request == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (reason != NULL, RTSP_EINVAL);
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (request != NULL, RTSP_EINVAL);
newmsg = g_new0 (RTSPMessage, 1);
*msg = newmsg;
return rtsp_message_init_response (code, reason, request, newmsg);
return rtsp_message_init_response (newmsg, code, reason, request);
}
RTSPResult
rtsp_message_init_response (RTSPStatusCode code, gchar * reason,
RTSPMessage * request, RTSPMessage * msg)
rtsp_message_init_response (RTSPMessage * msg, RTSPStatusCode code,
gchar * reason, RTSPMessage * request)
{
if (reason == NULL || msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (reason != NULL, RTSP_EINVAL);
rtsp_message_unset (msg);
msg->type = RTSP_MESSAGE_RESPONSE;
msg->type_data.response.code = code;
g_free (msg->type_data.response.reason);
msg->type_data.response.reason = g_strdup (reason);
if (msg->hdr_fields != NULL)
g_hash_table_destroy (msg->hdr_fields);
msg->hdr_fields =
g_hash_table_new_full (g_direct_hash, g_direct_equal, NULL, g_free);
if (msg->body) {
g_free (msg->body);
msg->body = NULL;
}
msg->body_size = 0;
if (request) {
/* FIXME copy headers */
}
@ -106,10 +91,11 @@ rtsp_message_init_response (RTSPStatusCode code, gchar * reason,
}
RTSPResult
rtsp_message_init_data (gint channel, RTSPMessage * msg)
rtsp_message_init_data (RTSPMessage * msg, gint channel)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
rtsp_message_unset (msg);
msg->type = RTSP_MESSAGE_DATA;
msg->type_data.data.channel = channel;
@ -117,13 +103,51 @@ rtsp_message_init_data (gint channel, RTSPMessage * msg)
return RTSP_OK;
}
RTSPResult
rtsp_message_unset (RTSPMessage * msg)
{
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
msg->type = RTSP_MESSAGE_INVALID;
msg->type_data.request.method = RTSP_INVALID;
g_free (msg->type_data.request.uri);
msg->type_data.request.uri = NULL;
msg->type_data.response.code = RTSP_STS_INVALID;
g_free (msg->type_data.response.reason);
msg->type_data.response.reason = NULL;
if (msg->hdr_fields != NULL)
g_hash_table_destroy (msg->hdr_fields);
msg->hdr_fields = NULL;
g_free (msg->body);
msg->body = NULL;
msg->body_size = 0;
return RTSP_OK;
}
RTSPResult
rtsp_message_free (RTSPMessage * msg)
{
RTSPResult res;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
res = rtsp_message_unset (msg);
if (res == RTSP_OK)
g_free (msg);
return res;
}
RTSPResult
rtsp_message_add_header (RTSPMessage * msg, RTSPHeaderField field,
gchar * value)
{
if (msg == NULL || value == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (value != NULL, RTSP_EINVAL);
g_hash_table_insert (msg->hdr_fields, GINT_TO_POINTER (field),
g_strdup (value));
@ -134,8 +158,7 @@ rtsp_message_add_header (RTSPMessage * msg, RTSPHeaderField field,
RTSPResult
rtsp_message_remove_header (RTSPMessage * msg, RTSPHeaderField field)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_hash_table_remove (msg->hdr_fields, GINT_TO_POINTER (field));
@ -148,8 +171,8 @@ rtsp_message_get_header (RTSPMessage * msg, RTSPHeaderField field,
{
gchar *val;
if (msg == NULL || value == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (value != NULL, RTSP_EINVAL);
val = g_hash_table_lookup (msg->hdr_fields, GINT_TO_POINTER (field));
if (val == NULL)
@ -163,8 +186,7 @@ rtsp_message_get_header (RTSPMessage * msg, RTSPHeaderField field,
RTSPResult
rtsp_message_set_body (RTSPMessage * msg, guint8 * data, guint size)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
return rtsp_message_take_body (msg, g_memdup (data, size), size);
}
@ -172,8 +194,7 @@ rtsp_message_set_body (RTSPMessage * msg, guint8 * data, guint size)
RTSPResult
rtsp_message_take_body (RTSPMessage * msg, guint8 * data, guint size)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
if (msg->body)
g_free (msg->body);
@ -187,8 +208,9 @@ rtsp_message_take_body (RTSPMessage * msg, guint8 * data, guint size)
RTSPResult
rtsp_message_get_body (RTSPMessage * msg, guint8 ** data, guint * size)
{
if (msg == NULL || data == NULL || size == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (data != NULL, RTSP_EINVAL);
g_return_val_if_fail (size != NULL, RTSP_EINVAL);
*data = msg->body;
*size = msg->body_size;
@ -199,8 +221,9 @@ rtsp_message_get_body (RTSPMessage * msg, guint8 ** data, guint * size)
RTSPResult
rtsp_message_steal_body (RTSPMessage * msg, guint8 ** data, guint * size)
{
if (msg == NULL || data == NULL || size == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (data != NULL, RTSP_EINVAL);
g_return_val_if_fail (size != NULL, RTSP_EINVAL);
*data = msg->body;
*size = msg->body_size;
@ -257,8 +280,7 @@ rtsp_message_dump (RTSPMessage * msg)
guint8 *data;
guint size;
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
if (msg->type == RTSP_MESSAGE_REQUEST) {
g_print ("request message %p\n", msg);

16
gst/rtsp/rtspmessage.h
View File

@ -28,6 +28,7 @@ G_BEGIN_DECLS
typedef enum
{
RTSP_MESSAGE_INVALID,
RTSP_MESSAGE_REQUEST,
RTSP_MESSAGE_RESPONSE,
RTSP_MESSAGE_DATA,
@ -58,15 +59,16 @@ typedef struct _RTSPMessage
} RTSPMessage;
RTSPResult rtsp_message_new_request (RTSPMethod method, gchar *uri, RTSPMessage **msg);
RTSPResult rtsp_message_init_request (RTSPMethod method, gchar *uri, RTSPMessage *msg);
RTSPResult rtsp_message_new_request (RTSPMessage **msg, RTSPMethod method, gchar *uri);
RTSPResult rtsp_message_init_request (RTSPMessage *msg, RTSPMethod method, gchar *uri);
RTSPResult rtsp_message_new_response (RTSPStatusCode code, gchar *reason,
RTSPMessage *request, RTSPMessage **msg);
RTSPResult rtsp_message_init_response (RTSPStatusCode code, gchar *reason,
RTSPMessage *request, RTSPMessage *msg);
RTSPResult rtsp_message_init_data (gint channel, RTSPMessage *msg);
RTSPResult rtsp_message_new_response (RTSPMessage **msg, RTSPStatusCode code, gchar *reason,
RTSPMessage *request);
RTSPResult rtsp_message_init_response (RTSPMessage *msg, RTSPStatusCode code, gchar *reason,
RTSPMessage *request);
RTSPResult rtsp_message_init_data (RTSPMessage *msg, gint channel);
RTSPResult rtsp_message_unset (RTSPMessage *msg);
RTSPResult rtsp_message_free (RTSPMessage *msg);

11
gst/rtsp/rtsptransport.c
View File

@ -27,8 +27,7 @@ rtsp_transport_new (RTSPTransport ** transport)
{
RTSPTransport *trans;
if (transport == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
trans = g_new0 (RTSPTransport, 1);
@ -40,6 +39,8 @@ rtsp_transport_new (RTSPTransport ** transport)
RTSPResult
rtsp_transport_init (RTSPTransport * transport)
{
g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
g_free (transport->destination);
g_free (transport->source);
g_free (transport->ssrc);
@ -83,8 +84,8 @@ rtsp_transport_parse (gchar * str, RTSPTransport * transport)
gchar **split, *down;
gint i;
if (str == NULL || transport == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
g_return_val_if_fail (str != NULL, RTSP_EINVAL);
rtsp_transport_init (transport);
@ -141,6 +142,8 @@ rtsp_transport_parse (gchar * str, RTSPTransport * transport)
RTSPResult
rtsp_transport_free (RTSPTransport * transport)
{
g_return_val_if_fail (transport != NULL, RTSP_EINVAL);
rtsp_transport_init (transport);
g_free (transport);
return RTSP_OK;

6
gst/rtsp/rtspurl.c
View File

@ -34,10 +34,10 @@ rtsp_url_parse (const gchar * urlstr, RTSPUrl ** url)
RTSPUrl *res;
gchar *p, *slash, *at, *col;
res = g_new0 (RTSPUrl, 1);
g_return_val_if_fail (urlstr != NULL, RTSP_EINVAL);
g_return_val_if_fail (url != NULL, RTSP_EINVAL);
if (urlstr == NULL)
goto invalid;
res = g_new0 (RTSPUrl, 1);
p = (gchar *) urlstr;
if (g_str_has_prefix (p, RTSP_PROTO)) {

29
gst/rtsp/sdpmessage.c
View File

@ -76,8 +76,7 @@ sdp_message_new (SDPMessage ** msg)
{
SDPMessage *newmsg;
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
newmsg = g_new0 (SDPMessage, 1);
@ -89,8 +88,7 @@ sdp_message_new (SDPMessage ** msg)
RTSPResult
sdp_message_init (SDPMessage * msg)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
FREE_STRING (msg->version);
FREE_STRING (msg->origin.username);
@ -123,8 +121,7 @@ sdp_message_init (SDPMessage * msg)
RTSPResult
sdp_message_clean (SDPMessage * msg)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
FREE_ARRAY (msg->emails);
FREE_ARRAY (msg->phones);
@ -140,8 +137,7 @@ sdp_message_clean (SDPMessage * msg)
RTSPResult
sdp_message_free (SDPMessage * msg)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
sdp_message_clean (msg);
@ -156,8 +152,7 @@ sdp_media_new (SDPMedia ** media)
{
SDPMedia *newmedia;
if (media == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (media != NULL, RTSP_EINVAL);
newmedia = g_new0 (SDPMedia, 1);
@ -169,8 +164,7 @@ sdp_media_new (SDPMedia ** media)
RTSPResult
sdp_media_init (SDPMedia * media)
{
if (media == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (media != NULL, RTSP_EINVAL);
FREE_STRING (media->media);
media->port = 0;
@ -578,8 +572,9 @@ sdp_message_parse_buffer (guint8 * data, guint size, SDPMessage * msg)
gchar buffer[4096];
gint idx = 0;
if (msg == NULL || data == NULL || size == 0)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_return_val_if_fail (data != NULL, RTSP_EINVAL);
g_return_val_if_fail (size != 0, RTSP_EINVAL);
c.state = SDP_SESSION;
c.msg = msg;
@ -651,8 +646,7 @@ print_media (SDPMedia * media)
RTSPResult
sdp_message_dump (SDPMessage * msg)
{
if (msg == NULL)
return RTSP_EINVAL;
g_return_val_if_fail (msg != NULL, RTSP_EINVAL);
g_print ("sdp packet %p:\n", msg);
g_print (" version: '%s'\n", msg->version);
@ -666,6 +660,7 @@ sdp_message_dump (SDPMessage * msg)
g_print (" session_name: '%s'\n", msg->session_name);
g_print (" information: '%s'\n", msg->information);
g_print (" uri: '%s'\n", msg->uri);
if (msg->emails->len > 0) {
gint i;
@ -710,7 +705,5 @@ sdp_message_dump (SDPMessage * msg)
print_media (&g_array_index (msg->medias, SDPMedia, i));
}
}
return RTSP_OK;
}