GstRTSPSession * session);
static void unlink_session_transports (GstRTSPClient * client,
GstRTSPSession * session, GstRTSPSessionMedia * sessmedia);
+static gboolean default_configure_client_media (GstRTSPClient * client,
+ GstRTSPMedia * media, GstRTSPStream * stream, GstRTSPContext * ctx);
static gboolean default_configure_client_transport (GstRTSPClient * client,
GstRTSPContext * ctx, GstRTSPTransport * ct);
static GstRTSPResult default_params_set (GstRTSPClient * client,
GstRTSPContext * ctx);
static GstRTSPResult default_params_get (GstRTSPClient * client,
GstRTSPContext * ctx);
-static gchar * default_make_path_from_uri (GstRTSPClient *client,
- const GstRTSPUrl *uri);
+static gchar *default_make_path_from_uri (GstRTSPClient * client,
+ const GstRTSPUrl * uri);
G_DEFINE_TYPE (GstRTSPClient, gst_rtsp_client, G_TYPE_OBJECT);
gobject_class->finalize = gst_rtsp_client_finalize;
klass->create_sdp = create_sdp;
+ klass->configure_client_media = default_configure_client_media;
klass->configure_client_transport = default_configure_client_transport;
klass->params_set = default_params_set;
klass->params_get = default_params_get;
GstRTSPUrl *uri;
GString *rtpinfo;
guint n_streams, i, infocount;
- gchar *str, *base_url;
+ gchar *str;
GstRTSPTimeRange *range;
GstRTSPResult res;
GstRTSPState rtspstate;
if (rtspstate != GST_RTSP_STATE_PLAYING && rtspstate != GST_RTSP_STATE_READY)
goto invalid_state;
+ /* in play we first unsuspend, media could be suspended from SDP or PAUSED */
+ if (!gst_rtsp_media_unsuspend (media))
+ goto unsuspend_failed;
+
/* parse the range header if we have one */
res = gst_rtsp_message_get_header (ctx->request, GST_RTSP_HDR_RANGE, &str, 0);
if (res == GST_RTSP_OK) {
/* 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;
stream = gst_rtsp_stream_transport_get_stream (trans);
if (gst_rtsp_stream_get_rtpinfo (stream, &rtptime, &seq)) {
- gchar *control;
+ const GstRTSPUrl *url;
+ gchar *url_str;
if (infocount > 0)
g_string_append (rtpinfo, ", ");
- 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);
+ url = gst_rtsp_stream_transport_get_url (trans);
+ url_str = gst_rtsp_url_get_request_uri (url);
+ g_string_append_printf (rtpinfo, "url=%s;seq=%u;rtptime=%u",
+ url_str, seq, rtptime);
+ g_free (url_str);
infocount++;
} else {
GST_WARNING ("RTP-Info cannot be determined for stream %d", i);
}
}
- g_free (base_url);
g_free (path);
/* construct the response now */
g_free (path);
return FALSE;
}
+unsuspend_failed:
+ {
+ GST_ERROR ("client %p: unsuspend failed", client);
+ send_generic_response (client, GST_RTSP_STS_SERVICE_UNAVAILABLE, ctx);
+ g_free (path);
+ return FALSE;
+ }
}
static void
}
static gboolean
-handle_blocksize (GstRTSPMedia * media, GstRTSPStream * stream,
- GstRTSPMessage * request)
+default_configure_client_media (GstRTSPClient * client, GstRTSPMedia * media,
+ GstRTSPStream * stream, GstRTSPContext * ctx)
{
+ GstRTSPMessage *request = ctx->request;
gchar *blocksize_str;
- gboolean ret = TRUE;
if (gst_rtsp_message_get_header (request, GST_RTSP_HDR_BLOCKSIZE,
&blocksize_str, 0) == GST_RTSP_OK) {
gchar *end;
blocksize = g_ascii_strtoull (blocksize_str, &end, 10);
- if (end == blocksize_str) {
- GST_ERROR ("failed to parse blocksize");
- ret = FALSE;
- } else {
- /* we don't want to change the mtu when this media
- * can be shared because it impacts other clients */
- if (gst_rtsp_media_is_shared (media))
- return TRUE;
-
- if (blocksize > G_MAXUINT)
- blocksize = G_MAXUINT;
- gst_rtsp_stream_set_mtu (stream, blocksize);
- }
+ if (end == blocksize_str)
+ goto parse_failed;
+
+ /* we don't want to change the mtu when this media
+ * can be shared because it impacts other clients */
+ if (gst_rtsp_media_is_shared (media))
+ goto done;
+
+ if (blocksize > G_MAXUINT)
+ blocksize = G_MAXUINT;
+
+ gst_rtsp_stream_set_mtu (stream, blocksize);
+ }
+done:
+ return TRUE;
+
+ /* ERRORS */
+parse_failed:
+ {
+ GST_ERROR_OBJECT (client, "failed to parse blocksize");
+ send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ return FALSE;
}
- return ret;
}
static gboolean
if (path[matched] == '\0')
goto control_not_found;
- /* path is what matched. We can modify the parsed uri in place */
+ /* path is what matched. */
path[matched] = '\0';
/* control is remainder */
control = &path[matched + 1];
ctx->sessmedia = sessmedia;
- /* set blocksize on this stream */
- if (!handle_blocksize (media, stream, ctx->request))
- goto invalid_blocksize;
+ if (!klass->configure_client_media (client, media, stream, ctx))
+ goto configure_media_failed_no_reply;
gst_rtsp_transport_new (&ct);
/* set in the session media transport */
trans = gst_rtsp_session_media_set_transport (sessmedia, stream, ct);
+ /* configure the url used to set this transport, this we will use when
+ * generating the response for the PLAY request */
+ gst_rtsp_stream_transport_set_url (trans, uri);
+
/* configure keepalive for this transport */
gst_rtsp_stream_transport_set_keepalive (trans,
(GstRTSPKeepAliveFunc) do_keepalive, session, NULL);
g_free (path);
return FALSE;
}
-invalid_blocksize:
+configure_media_failed_no_reply:
{
- GST_ERROR ("client %p: invalid blocksize", client);
- send_generic_response (client, GST_RTSP_STS_BAD_REQUEST, ctx);
+ GST_ERROR ("client %p: configure_media failed", client);
g_object_unref (session);
g_free (path);
+ /* error reply is already sent */
return FALSE;
}
unsupported_transports:
if (!(sdp = klass->create_sdp (client, media)))
goto no_sdp;
+ /* we suspend after the describe */
+ gst_rtsp_media_suspend (media);
g_object_unref (media);
gst_rtsp_message_init_response (ctx->response, GST_RTSP_STS_OK,
gst_rtsp_message_dump (request);
}
- GST_INFO ("client %p: received a request", client);
-
gst_rtsp_message_parse_request (request, &method, &uristr, &version);
+ GST_INFO ("client %p: received a request %s %s %s", client,
+ gst_rtsp_method_as_text (method), uristr,
+ gst_rtsp_version_as_text (version));
+
/* we can only handle 1.0 requests */
if (version != GST_RTSP_VERSION_1_0)
goto not_supported;
/* we always try to parse the url first */
if (strcmp (uristr, "*") == 0) {
/* special case where we have * as uri, keep uri = NULL */
- } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK)
- goto bad_request;
+ } else if (gst_rtsp_url_parse (uristr, &uri) != GST_RTSP_OK) {
+ /* check if the uristr is an absolute path <=> scheme and host information
+ * is missing */
+ gchar *scheme;
+
+ scheme = g_uri_parse_scheme (uristr);
+ if (scheme == NULL && g_str_has_prefix (uristr, "/")) {
+ gchar *absolute_uristr = NULL;
+
+ GST_WARNING_OBJECT (client, "request doesn't contain absolute url");
+ if (priv->server_ip == NULL) {
+ GST_WARNING_OBJECT (client, "host information missing");
+ goto bad_request;
+ }
+
+ absolute_uristr =
+ g_strdup_printf ("rtsp://%s%s", priv->server_ip, uristr);
+
+ GST_DEBUG_OBJECT (client, "absolute url: %s", absolute_uristr);
+ if (gst_rtsp_url_parse (absolute_uristr, &uri) != GST_RTSP_OK) {
+ g_free (absolute_uristr);
+ goto bad_request;
+ }
+ g_free (absolute_uristr);
+ } else {
+ g_free (scheme);
+ goto bad_request;
+ }
+ }
/* get the session if there is any */
res = gst_rtsp_message_get_header (request, GST_RTSP_HDR_SESSION, &sessid, 0);
* @context: (allow-none): a #GMainContext
*
* Attaches @client to @context. When the mainloop for @context is run, the
- * client will be dispatched. When @context is NULL, the default context will be
+ * client will be dispatched. When @context is %NULL, the default context will be
* used).
*
* This function should be called when the client properties and urls are fully
/**
* gst_rtsp_client_session_filter:
* @client: a #GstRTSPClient
- * @func: (scope call): a callback
+ * @func: (scope call) (allow-none): a callback
* @user_data: user data passed to @func
*
* Call @func for each session managed by @client. The result value of @func
* will also be added with an additional ref to the result #GList of this
* function..
*
+ * When @func is %NULL, #GST_RTSP_FILTER_REF will be assumed for each session.
+ *
* Returns: (element-type GstRTSPSession) (transfer full): a #GList with all
* sessions for which @func returned #GST_RTSP_FILTER_REF. After usage, each
* element in the #GList should be unreffed before the list is freed.
GList *result, *walk, *next;
g_return_val_if_fail (GST_IS_RTSP_CLIENT (client), NULL);
- g_return_val_if_fail (func != NULL, NULL);
priv = client->priv;
g_mutex_lock (&priv->lock);
for (walk = priv->sessions; walk; walk = next) {
GstRTSPSession *sess = walk->data;
+ GstRTSPFilterResult res;
next = g_list_next (walk);
- switch (func (client, sess, user_data)) {
+ if (func)
+ res = func (client, sess, user_data);
+ else
+ res = GST_RTSP_FILTER_REF;
+
+ switch (res) {
case GST_RTSP_FILTER_REMOVE:
/* stop watching the session and pretent it went away */
client_cleanup_session (client, sess);
projects / platform / upstream / gstreamer.git / blobdiff