GstRTSPMessage request = { 0 };
GstRTSPResult res;
GstRTSPMethod method;
+ gchar *control;
GST_DEBUG_OBJECT (src, "creating server keep-alive");
else
method = GST_RTSP_OPTIONS;
- res = gst_rtsp_message_init_request (&request, method, src->req_location);
+ if (src->control)
+ control = src->control;
+ else
+ control = src->req_location;
+
+ if (control == NULL)
+ goto no_control;
+
+ res = gst_rtsp_message_init_request (&request, method, control);
if (res < 0)
goto send_error;
return GST_RTSP_OK;
/* ERRORS */
+no_control:
+ {
+ GST_WARNING_OBJECT (src, "no control url to send keepalive");
+ return GST_RTSP_OK;
+ }
send_error:
{
gchar *str = gst_rtsp_strresult (res);
return TRUE;
}
+static gboolean
+gst_rtspsrc_from_sdp (GstRTSPSrc * src, guint8 * data, guint size)
+{
+ GstSDPMessage sdp = { 0 };
+ gint i, n_streams;
+
+ GST_DEBUG_OBJECT (src, "parse SDP...");
+ gst_sdp_message_init (&sdp);
+ gst_sdp_message_parse_buffer (data, size, &sdp);
+
+ if (src->debug)
+ gst_sdp_message_dump (&sdp);
+
+ gst_rtsp_ext_list_parse_sdp (src->extensions, &sdp, src->props);
+
+ /* parse range for duration reporting. */
+ {
+ const gchar *range;
+
+ for (i = 0;; i++) {
+ range = gst_sdp_message_get_attribute_val_n (&sdp, "range", i);
+ if (range == NULL)
+ break;
+
+ /* keep track of the range and configure it in the segment */
+ if (gst_rtspsrc_parse_range (src, range, &src->segment))
+ break;
+ }
+ }
+ /* try to find a global control attribute */
+ {
+ const gchar *control;
+
+ for (i = 0;; i++) {
+ control = gst_sdp_message_get_attribute_val_n (&sdp, "control", i);
+ if (control == NULL)
+ break;
+
+ /* only take fully qualified urls */
+ if (g_str_has_prefix (control, "rtsp://"))
+ break;
+ }
+ g_free (src->control);
+ src->control = g_strdup (control);
+ }
+
+ /* create streams */
+ n_streams = gst_sdp_message_medias_len (&sdp);
+ for (i = 0; i < n_streams; i++) {
+ gst_rtspsrc_create_stream (src, &sdp, i);
+ }
+
+ src->state = GST_RTSP_STATE_INIT;
+
+ /* setup streams */
+ if (!gst_rtspsrc_setup_streams (src))
+ goto setup_failed;
+
+ src->state = GST_RTSP_STATE_READY;
+
+ gst_sdp_message_uninit (&sdp);
+
+ return TRUE;
+
+setup_failed:
+ {
+ gst_sdp_message_uninit (&sdp);
+ return FALSE;
+ }
+}
+
static gboolean
gst_rtspsrc_open (GstRTSPSrc * src)
{
GstRTSPMessage response = { 0 };
guint8 *data;
guint size;
- gint i, n_streams;
- GstSDPMessage sdp = { 0 };
gchar *respcont = NULL;
GstRTSPUrl *url;
if (data == NULL)
goto no_describe;
- GST_DEBUG_OBJECT (src, "parse SDP...");
- gst_sdp_message_init (&sdp);
- gst_sdp_message_parse_buffer (data, size, &sdp);
-
- if (src->debug)
- gst_sdp_message_dump (&sdp);
-
- gst_rtsp_ext_list_parse_sdp (src->extensions, &sdp, src->props);
-
- /* parse range for duration reporting. */
- {
- const gchar *range;
-
- for (i = 0;; i++) {
- range = gst_sdp_message_get_attribute_val_n (&sdp, "range", i);
- if (range == NULL)
- break;
-
- /* keep track of the range and configure it in the segment */
- if (gst_rtspsrc_parse_range (src, range, &src->segment))
- break;
- }
- }
- /* try to find a global control attribute */
- g_free (src->control);
- src->control = NULL;
- {
- const gchar *control;
-
- for (i = 0;; i++) {
- control = gst_sdp_message_get_attribute_val_n (&sdp, "control", i);
- if (control == NULL)
- break;
-
- if (g_str_has_prefix (control, "rtsp://")) {
- src->control = g_strdup (control);
- break;
- }
- }
- }
-
- /* create streams */
- n_streams = gst_sdp_message_medias_len (&sdp);
- for (i = 0; i < n_streams; i++) {
- gst_rtspsrc_create_stream (src, &sdp, i);
- }
-
- src->state = GST_RTSP_STATE_INIT;
-
- /* setup streams */
- if (!gst_rtspsrc_setup_streams (src))
- goto setup_failed;
-
- src->state = GST_RTSP_STATE_READY;
- GST_RTSP_STATE_UNLOCK (src);
+ if (!gst_rtspsrc_from_sdp (src, data, size))
+ goto sdp_failed;
/* clean up any messages */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
- gst_sdp_message_uninit (&sdp);
return TRUE;
("Server can not provide an SDP."));
goto cleanup_error;
}
-setup_failed:
+sdp_failed:
{
gst_rtspsrc_close (src);
/* error was posted */
GST_RTSP_STATE_UNLOCK (src);
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
- gst_sdp_message_uninit (&sdp);
return FALSE;
}
}
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res;
+ GList *walk;
gboolean ret = FALSE;
gchar *control;
else
control = src->req_location;
- if (src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)) {
+ if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
+ goto not_supported;
+
+ for (walk = src->streams; walk; walk = g_list_next (walk)) {
+ GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+ gchar *setup_url;
+
+ /* try aggregate control first but do non-aggregate control otherwise */
+ if (control)
+ setup_url = control;
+ else if ((setup_url = stream->setup_url) == NULL)
+ continue;
+
/* do TEARDOWN */
- res = gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, control);
+ res =
+ gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
if (res < 0)
goto create_request_failed;
/* FIXME, parse result? */
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
- } else {
- GST_DEBUG_OBJECT (src,
- "TEARDOWN and PLAY not supported, can't do TEARDOWN");
+
+ /* early exit when we did aggregate control */
+ if (control)
+ break;
}
close:
ret = FALSE;
goto close;
}
+not_supported:
+ {
+ GST_DEBUG_OBJECT (src,
+ "TEARDOWN and PLAY not supported, can't do TEARDOWN");
+ goto close;
+ }
}
/* RTP-Info is of the format:
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
GstRTSPResult res;
+ GList *walk;
gchar *hval;
gfloat fval;
gint hval_idx;
if (!src->connection || !src->connected)
goto done;
- /* construct a control url */
- if (src->control)
- control = src->control;
- else
- control = src->req_location;
-
/* waiting for connection idle, we were flushing so any attempt at doing data
* transfer will result in pausing the tasks. */
GST_DEBUG_OBJECT (src, "wait for connection idle");
GST_DEBUG_OBJECT (src, "stop connection flush");
gst_rtsp_connection_flush (src->connection, FALSE);
- /* do play */
- res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, control);
- if (res < 0)
- goto create_request_failed;
+ /* construct a control url */
+ if (src->control)
+ control = src->control;
+ else
+ control = src->req_location;
- if (src->need_range) {
- hval = gen_range_header (src, segment);
+ for (walk = src->streams; walk; walk = g_list_next (walk)) {
+ GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+ gchar *setup_url;
- gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
- g_free (hval);
- src->need_range = FALSE;
- }
+ /* try aggregate control first but do non-aggregate control otherwise */
+ if (control)
+ setup_url = control;
+ else if ((setup_url = stream->setup_url) == NULL)
+ continue;
- if (segment->rate != 1.0) {
- hval = gst_rtspsrc_dup_printf ("%f", segment->rate);
- if (src->skip)
- gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
- else
- gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
- g_free (hval);
- }
+ /* do play */
+ res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
+ if (res < 0)
+ goto create_request_failed;
- if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
- goto send_error;
+ if (src->need_range) {
+ hval = gen_range_header (src, segment);
- gst_rtsp_message_unset (&request);
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_RANGE, hval);
+ g_free (hval);
+ src->need_range = FALSE;
+ }
- /* parse RTP npt field. This is the current position in the stream (Normal
- * Play Time) and should be put in the NEWSEGMENT position field. */
- if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
- 0) == GST_RTSP_OK)
- gst_rtspsrc_parse_range (src, hval, segment);
-
- /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
- segment->rate = 1.0;
-
- /* parse Speed header. This is the intended playback rate of the stream
- * and should be put in the NEWSEGMENT rate field. */
- if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
- 0) == GST_RTSP_OK) {
- if (gst_rtspsrc_get_float (hval, &fval) > 0)
- segment->rate = fval;
- } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE, &hval,
- 0) == GST_RTSP_OK) {
- if (gst_rtspsrc_get_float (hval, &fval) > 0)
- segment->rate = fval;
- }
-
- /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
- * for the RTP packets. If this is not present, we assume all starts from 0...
- * This is info for the RTP session manager that we pass to it in caps. */
- hval_idx = 0;
- while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
- &hval, hval_idx++) == GST_RTSP_OK)
- gst_rtspsrc_parse_rtpinfo (src, hval);
+ if (segment->rate != 1.0) {
+ hval = gst_rtspsrc_dup_printf ("%f", segment->rate);
+ if (src->skip)
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
+ else
+ gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
+ g_free (hval);
+ }
- gst_rtsp_message_unset (&response);
+ if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
+ goto send_error;
+
+ gst_rtsp_message_unset (&request);
+
+ /* parse RTP npt field. This is the current position in the stream (Normal
+ * Play Time) and should be put in the NEWSEGMENT position field. */
+ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
+ 0) == GST_RTSP_OK)
+ gst_rtspsrc_parse_range (src, hval, segment);
+
+ /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
+ segment->rate = 1.0;
+
+ /* parse Speed header. This is the intended playback rate of the stream
+ * and should be put in the NEWSEGMENT rate field. */
+ if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
+ 0) == GST_RTSP_OK) {
+ if (gst_rtspsrc_get_float (hval, &fval) > 0)
+ segment->rate = fval;
+ } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
+ &hval, 0) == GST_RTSP_OK) {
+ if (gst_rtspsrc_get_float (hval, &fval) > 0)
+ segment->rate = fval;
+ }
+
+ /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
+ * for the RTP packets. If this is not present, we assume all starts from 0...
+ * This is info for the RTP session manager that we pass to it in caps. */
+ hval_idx = 0;
+ while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
+ &hval, hval_idx++) == GST_RTSP_OK)
+ gst_rtspsrc_parse_rtpinfo (src, hval);
+
+ gst_rtsp_message_unset (&response);
+
+ /* early exit when we did aggregate control */
+ if (control)
+ break;
+ }
/* configure the caps of the streams after we parsed all headers. */
gst_rtspsrc_configure_caps (src, segment);
{
GstRTSPMessage request = { 0 };
GstRTSPMessage response = { 0 };
+ GList *walk;
gchar *control;
GST_RTSP_STATE_LOCK (src);
else
control = src->req_location;
- /* do pause */
- if (gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE, control) < 0)
- goto create_request_failed;
+ /* loop over the streams. We might exit the loop early when we could do an
+ * aggregate control */
+ for (walk = src->streams; walk; walk = g_list_next (walk)) {
+ GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+ gchar *setup_url;
- if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
- goto send_error;
+ /* try aggregate control first but do non-aggregate control otherwise */
+ if (control)
+ setup_url = control;
+ else if ((setup_url = stream->setup_url) == NULL)
+ continue;
- gst_rtsp_message_unset (&request);
- gst_rtsp_message_unset (&response);
+ if (gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE, setup_url) < 0)
+ goto create_request_failed;
+
+ if (gst_rtspsrc_send (src, &request, &response, NULL) < 0)
+ goto send_error;
+
+ gst_rtsp_message_unset (&request);
+ gst_rtsp_message_unset (&response);
+
+ /* exit early when we did agregate control */
+ if (control)
+ break;
+ }
if (idle && src->task) {
GST_DEBUG_OBJECT (src, "starting idle task again");
projects / platform / upstream / gstreamer.git / commitdiff