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gst/rtsp/gstrtspsrc.c: Refactor the udp and interleaved loop function a bit.

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Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop):
Refactor the udp and interleaved loop function a bit.
This commit is contained in:
Wim Taymans 2007-08-18 19:44:55 +00:00
parent 0dcafb0635
commit 60bf53248b
2 changed files with 55 additions and 84 deletions
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7
ChangeLog
View file

@ -1,3 +1,10 @@
2007-08-18 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop):
Refactor the udp and interleaved loop function a bit.
2007-08-17 Wim Taymans <wim.taymans@gmail.com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),

132
gst/rtsp/gstrtspsrc.c
View file

@ -2314,7 +2314,7 @@ send_error:
}
}
static void
static GstFlowReturn
gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
{
GstRTSPMessage message = { 0 };
@ -2459,17 +2459,15 @@ gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
if (!is_rtcp) {
/* combine all stream flows for the data transport */
ret = gst_rtspsrc_combine_flows (src, stream, ret);
if (ret != GST_FLOW_OK)
goto need_pause;
}
return;
return ret;
/* ERRORS */
unknown_stream:
{
GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
gst_rtsp_message_unset (&message);
return;
return GST_FLOW_OK;
}
timeout:
{
@ -2477,16 +2475,14 @@ timeout:
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Timeout while waiting for server message."));
gst_rtsp_message_unset (&message);
ret = GST_FLOW_UNEXPECTED;
goto need_pause;
return GST_FLOW_UNEXPECTED;
}
server_eof:
{
GST_DEBUG_OBJECT (src, "we got an eof from the server");
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("The server closed the connection."));
ret = GST_FLOW_UNEXPECTED;
goto need_pause;
return GST_FLOW_UNEXPECTED;
}
interrupt:
{
@ -2494,8 +2490,7 @@ interrupt:
GST_DEBUG_OBJECT (src, "got interrupted: stop connection flush");
/* unset flushing so we can do something else */
gst_rtsp_connection_flush (src->connection, FALSE);
ret = GST_FLOW_WRONG_STATE;
goto need_pause;
return GST_FLOW_WRONG_STATE;
}
receive_error:
{
@ -2506,8 +2501,7 @@ receive_error:
g_free (str);
gst_rtsp_message_unset (&message);
ret = GST_FLOW_ERROR;
goto need_pause;
return GST_FLOW_ERROR;
}
handle_request_failed:
{
@ -2517,52 +2511,22 @@ handle_request_failed:
("Could not handle server message. (%s)", str));
g_free (str);
gst_rtsp_message_unset (&message);
ret = GST_FLOW_ERROR;
goto need_pause;
return GST_FLOW_ERROR;
}
invalid_length:
{
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Short message received, ignoring."));
gst_rtsp_message_unset (&message);
return;
}
need_pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
src->running = FALSE;
gst_task_pause (src->task);
if (GST_FLOW_IS_FATAL (ret) || ret == GST_FLOW_NOT_LINKED) {
if (ret == GST_FLOW_UNEXPECTED) {
/* perform EOS logic */
if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
gst_element_post_message (GST_ELEMENT_CAST (src),
gst_message_new_segment_done (GST_OBJECT_CAST (src),
src->segment.format, src->segment.last_stop));
} else {
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
} else {
/* for fatal errors we post an error message, post the error before the
* EOS so the app knows about the error first. */
GST_ELEMENT_ERROR (src, STREAM, FAILED,
("Internal data flow error."),
("streaming task paused, reason %s (%d)", reason, ret));
gst_rtspsrc_push_event (src, gst_event_new_eos ());
}
}
return;
return GST_FLOW_OK;
}
}
static void
static GstFlowReturn
gst_rtspsrc_loop_udp (GstRTSPSrc * src)
{
gboolean restart = FALSE;
GstRTSPResult res;
GstFlowReturn ret = GST_FLOW_OK;
GST_OBJECT_LOCK (src);
if (src->loop_cmd == CMD_STOP)
@ -2697,20 +2661,14 @@ gst_rtspsrc_loop_udp (GstRTSPSrc * src)
goto play_failed;
done:
return;
return GST_FLOW_OK;
/* ERRORS */
stopping:
{
GST_DEBUG_OBJECT (src, "we are stopping");
GST_OBJECT_UNLOCK (src);
ret = GST_FLOW_WRONG_STATE;
goto need_pause;
}
{
GST_DEBUG_OBJECT (src, "we got an eof from the server");
ret = GST_FLOW_UNEXPECTED;
goto need_pause;
return GST_FLOW_WRONG_STATE;
}
receive_error:
{
@ -2719,8 +2677,7 @@ receive_error:
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not receive message. (%s)", str));
g_free (str);
ret = GST_FLOW_ERROR;
goto need_pause;
return GST_FLOW_ERROR;
}
handle_request_failed:
{
@ -2729,8 +2686,7 @@ handle_request_failed:
GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
("Could not handle server message. (%s)", str));
g_free (str);
ret = GST_FLOW_ERROR;
goto need_pause;
return GST_FLOW_ERROR;
}
no_protocols:
{
@ -2738,20 +2694,49 @@ no_protocols:
/* no transport possible, post an error and stop */
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Could not connect to server, no protocols left"));
ret = GST_FLOW_ERROR;
goto need_pause;
return GST_FLOW_ERROR;
}
open_failed:
{
GST_DEBUG_OBJECT (src, "open failed");
return;
return GST_FLOW_OK;
}
play_failed:
{
GST_DEBUG_OBJECT (src, "play failed");
return;
return GST_FLOW_OK;
}
need_pause:
}
static void
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
{
GST_OBJECT_LOCK (src);
src->loop_cmd = cmd;
if (flush) {
GST_DEBUG_OBJECT (src, "start connection flush");
gst_rtsp_connection_flush (src->connection, TRUE);
}
GST_OBJECT_UNLOCK (src);
}
static void
gst_rtspsrc_loop (GstRTSPSrc * src)
{
GstFlowReturn ret;
if (src->interleaved)
ret = gst_rtspsrc_loop_interleaved (src);
else
ret = gst_rtspsrc_loop_udp (src);
if (ret != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
pause:
{
const gchar *reason = gst_flow_get_name (ret);
@ -2781,27 +2766,6 @@ need_pause:
}
}
static void
gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gboolean flush)
{
GST_OBJECT_LOCK (src);
src->loop_cmd = cmd;
if (flush) {
GST_DEBUG_OBJECT (src, "start connection flush");
gst_rtsp_connection_flush (src->connection, TRUE);
}
GST_OBJECT_UNLOCK (src);
}
static void
gst_rtspsrc_loop (GstRTSPSrc * src)
{
if (src->interleaved)
gst_rtspsrc_loop_interleaved (src);
else
gst_rtspsrc_loop_udp (src);
}
#ifndef GST_DISABLE_GST_DEBUG
const gchar *
gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)