mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-18 14:26:43 +00:00
Code Issues 1 Projects Releases Wiki Activity

client: Fix RTPInfo header

Browse source 
Refactor the method to make the content_base.
Use the content-base and the control url to construct the RTPInfo
url.
This commit is contained in:
Wim Taymans 2013-09-17 14:39:44 +02:00
parent 4d6c038fc7
commit b41422bad7
1 changed files with 43 additions and 38 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

81
gst/rtsp-server/rtsp-client.c
View file

@ -978,6 +978,28 @@ invalid_state:
}
}
/* convert @url and @path to a URL used as a content base for the factory
* located at @path */
static gchar *
make_base_url (GstRTSPClient * client, GstRTSPUrl * url, gchar * path)
{
GstRTSPUrl tmp;
gchar *result, *trail;
/* check for trailing '/' and append one */
trail = (path[strlen (path) - 1] != '/' ? "/" : "");
tmp = *url;
tmp.user = NULL;
tmp.passwd = NULL;
tmp.abspath = g_strdup_printf ("%s%s", path, trail);
tmp.query = NULL;
result = gst_rtsp_url_get_request_uri (&tmp);
g_free (tmp.abspath);
return result;
}
static gboolean
handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
{
@ -985,23 +1007,24 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
GstRTSPSessionMedia *sessmedia;
GstRTSPMedia *media;
GstRTSPStatusCode code;
GstRTSPUrl *uri;
GString *rtpinfo;
guint n_streams, i, infocount;
gchar *str;
gchar *str, *base_url;
GstRTSPTimeRange *range;
GstRTSPResult res;
GstRTSPState rtspstate;
GstRTSPRangeUnit unit = GST_RTSP_RANGE_NPT;
const gchar *path;
gchar *path;
gint matched;
if (!(session = ctx->session))
goto no_session;
if (!ctx->uri)
if (!(uri = ctx->uri))
goto no_uri;
path = ctx->uri->abspath;
path = uri->abspath;
/* get a handle to the configuration of the media in the session */
sessmedia = gst_rtsp_session_get_media (session, path, &matched);
@ -1033,12 +1056,13 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
/* grab RTPInfo from the payloaders now */
rtpinfo = g_string_new ("");
base_url = make_base_url (client, uri, path);
n_streams = gst_rtsp_media_n_streams (media);
for (i = 0, infocount = 0; i < n_streams; i++) {
GstRTSPStreamTransport *trans;
GstRTSPStream *stream;
const GstRTSPTransport *tr;
gchar *uristr;
guint rtptime, seq;
/* get the transport, if there is no transport configured, skip this stream */
@ -1056,19 +1080,22 @@ handle_play_request (GstRTSPClient * client, GstRTSPContext * ctx)
stream = gst_rtsp_stream_transport_get_stream (trans);
if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
gchar *control;
if (infocount > 0)
g_string_append (rtpinfo, ", ");
uristr = gst_rtsp_url_get_request_uri (ctx->uri);
g_string_append_printf (rtpinfo, "url=%s/stream=%d;seq=%u;rtptime=%u",
uristr, i, seq, rtptime);
g_free (uristr);
control = gst_rtsp_stream_get_control (stream);
g_string_append_printf (rtpinfo, "url=%s%s;seq=%u;rtptime=%u",
base_url, control, seq, rtptime);
g_free (control);
infocount++;
} else {
GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
}
}
g_free (base_url);
/* construct the response now */
code = GST_RTSP_STS_OK;
@ -1624,8 +1651,8 @@ handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
GstRTSPClientPrivate *priv = client->priv;
GstRTSPResult res;
GstSDPMessage *sdp;
guint i, str_len;
gchar *path, *str, *str_query, *content_base;
guint i;
gchar *path, *str;
GstRTSPMedia *media;
GstRTSPClientClass *klass;
@ -1659,8 +1686,6 @@ handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
if (!(media = find_media (client, ctx, path, NULL)))
goto no_media;
g_free (path);
/* create an SDP for the media object on this client */
if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
@ -1674,32 +1699,11 @@ handle_describe_request (GstRTSPClient * client, GstRTSPContext * ctx)
"application/sdp");
/* content base for some clients that might screw up creating the setup uri */
str = gst_rtsp_url_get_request_uri (ctx->uri);
str_len = strlen (str);
str = make_base_url (client, ctx->uri, path);
g_free (path);
/* check for query part */
if (ctx->uri->query != NULL) {
str_query = g_strrstr (str, "?");
*str_query = '\0';
str_len = strlen (str);
}
/* check for trailing '/' and append one */
if (str[str_len - 1] != '/') {
content_base = g_malloc (str_len + 2);
memcpy (content_base, str, str_len);
content_base[str_len] = '/';
content_base[str_len + 1] = '\0';
g_free (str);
} else {
content_base = str;
}
GST_INFO ("adding content-base: %s", content_base);
gst_rtsp_message_add_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE,
content_base);
g_free (content_base);
GST_INFO ("adding content-base: %s", str);
gst_rtsp_message_take_header (ctx->response, GST_RTSP_HDR_CONTENT_BASE, str);
/* add SDP to the response body */
str = gst_sdp_message_as_text (sdp);
@ -1743,6 +1747,7 @@ no_sdp:
{
GST_ERROR ("client %p: can't create SDP", client);
send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
g_free (path);
g_object_unref (media);
return FALSE;
}