mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
rtspsrc: expose and implement is-live property
This is useful to support the ONVIF case: when is-live is set to FALSE and onvif-rate-control is no, the client can control the rate of delivery and arrange for the server to block and still keep sending when unblocked, without requiring back and forth PAUSE / PLAY requests. This enables, amongst other things, fast frame stepping on the client side. When is-live is FALSE, we don't use a manager at all. This case was actually already pretty well handled by the current code. The standard manager, rtpbin, is simply no longer needed in this case. Applications can instantiate a downloadbuffer after rtspsrc if needed.
This commit is contained in:
parent
75f53631e5
commit
5c7423d73c
2 changed files with 92 additions and 20 deletions
|
@ -282,6 +282,7 @@ gst_rtsp_backchannel_get_type (void)
|
|||
#define DEFAULT_TEARDOWN_TIMEOUT (100 * GST_MSECOND)
|
||||
#define DEFAULT_ONVIF_MODE FALSE
|
||||
#define DEFAULT_ONVIF_RATE_CONTROL TRUE
|
||||
#define DEFAULT_IS_LIVE TRUE
|
||||
|
||||
enum
|
||||
{
|
||||
|
@ -328,7 +329,8 @@ enum
|
|||
PROP_BACKCHANNEL,
|
||||
PROP_TEARDOWN_TIMEOUT,
|
||||
PROP_ONVIF_MODE,
|
||||
PROP_ONVIF_RATE_CONTROL
|
||||
PROP_ONVIF_RATE_CONTROL,
|
||||
PROP_IS_LIVE
|
||||
};
|
||||
|
||||
#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
|
||||
|
@ -977,6 +979,22 @@ gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
|
|||
DEFAULT_ONVIF_RATE_CONTROL,
|
||||
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
||||
|
||||
/**
|
||||
* GstRtspSrc:is-live
|
||||
*
|
||||
* Whether to act as a live source. This is useful in combination with
|
||||
* #GstRtspSrc:onvif-rate-control set to %FALSE and usage of the TCP
|
||||
* protocol. In that situation, data delivery rate can be entirely
|
||||
* controlled from the client side, enabling features such as frame
|
||||
* stepping and instantaneous rate changes.
|
||||
*
|
||||
* Since: 1.18
|
||||
*/
|
||||
g_object_class_install_property (gobject_class, PROP_IS_LIVE,
|
||||
g_param_spec_boolean ("is-live", "Is live",
|
||||
"Whether to act as a live source",
|
||||
DEFAULT_IS_LIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
||||
|
||||
/**
|
||||
* GstRTSPSrc::handle-request:
|
||||
* @rtspsrc: a #GstRTSPSrc
|
||||
|
@ -1379,6 +1397,7 @@ gst_rtspsrc_init (GstRTSPSrc * src)
|
|||
src->teardown_timeout = DEFAULT_TEARDOWN_TIMEOUT;
|
||||
src->onvif_mode = DEFAULT_ONVIF_MODE;
|
||||
src->onvif_rate_control = DEFAULT_ONVIF_RATE_CONTROL;
|
||||
src->is_live = DEFAULT_IS_LIVE;
|
||||
|
||||
/* get a list of all extensions */
|
||||
src->extensions = gst_rtsp_ext_list_get ();
|
||||
|
@ -1723,6 +1742,9 @@ gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
|
|||
case PROP_ONVIF_RATE_CONTROL:
|
||||
rtspsrc->onvif_rate_control = g_value_get_boolean (value);
|
||||
break;
|
||||
case PROP_IS_LIVE:
|
||||
rtspsrc->is_live = g_value_get_boolean (value);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
|
@ -1893,6 +1915,9 @@ gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
|
|||
case PROP_ONVIF_RATE_CONTROL:
|
||||
g_value_set_boolean (value, rtspsrc->onvif_rate_control);
|
||||
break;
|
||||
case PROP_IS_LIVE:
|
||||
g_value_set_boolean (value, rtspsrc->is_live);
|
||||
break;
|
||||
default:
|
||||
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
||||
break;
|
||||
|
@ -2853,11 +2878,14 @@ gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
|
|||
stream->discont = TRUE;
|
||||
}
|
||||
|
||||
/* and continue playing if needed */
|
||||
/* and continue playing if needed. If we are not acting as a live source,
|
||||
* then only the RTSP PLAYING state, set earlier, matters. */
|
||||
GST_OBJECT_LOCK (src);
|
||||
playing = (GST_STATE_PENDING (src) == GST_STATE_VOID_PENDING
|
||||
&& GST_STATE (src) == GST_STATE_PLAYING)
|
||||
|| (GST_STATE_PENDING (src) == GST_STATE_PLAYING);
|
||||
if (src->is_live) {
|
||||
playing = (GST_STATE_PENDING (src) == GST_STATE_VOID_PENDING
|
||||
&& GST_STATE (src) == GST_STATE_PLAYING)
|
||||
|| (GST_STATE_PENDING (src) == GST_STATE_PLAYING);
|
||||
}
|
||||
GST_OBJECT_UNLOCK (src);
|
||||
|
||||
if (src->version >= GST_RTSP_VERSION_2_0) {
|
||||
|
@ -3029,7 +3057,7 @@ gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
|
|||
{
|
||||
/* we are live with a min latency of 0 and unlimited max latency, this
|
||||
* result will be updated by the session manager if there is any. */
|
||||
gst_query_set_latency (query, TRUE, 0, -1);
|
||||
gst_query_set_latency (query, src->is_live, 0, -1);
|
||||
break;
|
||||
}
|
||||
default:
|
||||
|
@ -3502,7 +3530,10 @@ on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
|
|||
"stream-number", G_TYPE_INT, stream->id, "ssrc", G_TYPE_UINT,
|
||||
stream->ssrc, NULL)));
|
||||
|
||||
on_timeout_common (session, source, stream);
|
||||
/* In non-live mode, timeouts can occur if we are PAUSED, this doesn't mean
|
||||
* the stream is EOS, it may simply be blocked */
|
||||
if (src->is_live || !src->interleaved)
|
||||
on_timeout_common (session, source, stream);
|
||||
}
|
||||
|
||||
static void
|
||||
|
@ -3830,6 +3861,9 @@ gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
|
|||
gchar *name;
|
||||
GstStateChangeReturn ret;
|
||||
|
||||
if (!src->is_live)
|
||||
goto use_no_manager;
|
||||
|
||||
/* find a manager */
|
||||
if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
|
||||
goto no_manager;
|
||||
|
@ -5338,6 +5372,11 @@ gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
|
|||
GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
|
||||
GstCaps *caps;
|
||||
|
||||
/* Activate in advance so that the stream-start event is registered */
|
||||
if (stream->srcpad) {
|
||||
gst_pad_set_active (stream->srcpad, TRUE);
|
||||
}
|
||||
|
||||
stream_id =
|
||||
g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
|
||||
event = gst_event_new_stream_start (stream_id);
|
||||
|
@ -7986,6 +8025,12 @@ gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
|
|||
if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
|
||||
goto open_failed;
|
||||
|
||||
if (src->initial_seek) {
|
||||
if (!gst_rtspsrc_perform_seek (src, src->initial_seek))
|
||||
goto initial_seek_failed;
|
||||
gst_event_replace (&src->initial_seek, NULL);
|
||||
}
|
||||
|
||||
done:
|
||||
if (async)
|
||||
gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
|
||||
|
@ -8005,6 +8050,13 @@ open_failed:
|
|||
src->open_error = TRUE;
|
||||
goto done;
|
||||
}
|
||||
initial_seek_failed:
|
||||
{
|
||||
GST_WARNING_OBJECT (src, "Failed to perform initial seek");
|
||||
ret = GST_RTSP_ERROR;
|
||||
src->open_error = TRUE;
|
||||
goto done;
|
||||
}
|
||||
}
|
||||
|
||||
static GstRTSPResult
|
||||
|
@ -9028,17 +9080,22 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
|
|||
/* first attempt, don't ignore timeouts */
|
||||
rtspsrc->ignore_timeout = FALSE;
|
||||
rtspsrc->open_error = FALSE;
|
||||
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
|
||||
if (rtspsrc->is_live)
|
||||
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
|
||||
else
|
||||
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
set_manager_buffer_mode (rtspsrc);
|
||||
/* fall-through */
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
/* unblock the tcp tasks and make the loop waiting */
|
||||
if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
|
||||
/* make sure it is waiting before we send PAUSE or PLAY below */
|
||||
GST_RTSP_STREAM_LOCK (rtspsrc);
|
||||
GST_RTSP_STREAM_UNLOCK (rtspsrc);
|
||||
if (rtspsrc->is_live) {
|
||||
/* unblock the tcp tasks and make the loop waiting */
|
||||
if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
|
||||
/* make sure it is waiting before we send PAUSE or PLAY below */
|
||||
GST_RTSP_STREAM_LOCK (rtspsrc);
|
||||
GST_RTSP_STREAM_UNLOCK (rtspsrc);
|
||||
}
|
||||
}
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
|
@ -9056,16 +9113,22 @@ gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
|
|||
ret = GST_STATE_CHANGE_SUCCESS;
|
||||
break;
|
||||
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
||||
ret = GST_STATE_CHANGE_NO_PREROLL;
|
||||
if (rtspsrc->is_live)
|
||||
ret = GST_STATE_CHANGE_NO_PREROLL;
|
||||
else
|
||||
ret = GST_STATE_CHANGE_SUCCESS;
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
||||
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
|
||||
if (rtspsrc->is_live)
|
||||
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
|
||||
ret = GST_STATE_CHANGE_SUCCESS;
|
||||
break;
|
||||
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
||||
/* send pause request and keep the idle task around */
|
||||
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
|
||||
ret = GST_STATE_CHANGE_NO_PREROLL;
|
||||
if (rtspsrc->is_live) {
|
||||
/* send pause request and keep the idle task around */
|
||||
gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
|
||||
}
|
||||
ret = GST_STATE_CHANGE_SUCCESS;
|
||||
break;
|
||||
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
||||
gst_rtspsrc_loop_send_cmd_and_wait (rtspsrc, CMD_CLOSE, CMD_ALL,
|
||||
|
@ -9109,8 +9172,14 @@ gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
|
|||
rtspsrc = GST_RTSPSRC (element);
|
||||
|
||||
if (GST_EVENT_TYPE (event) == GST_EVENT_SEEK) {
|
||||
res = gst_rtspsrc_perform_seek (rtspsrc, event);
|
||||
gst_event_unref (event);
|
||||
if (rtspsrc->state >= GST_RTSP_STATE_READY) {
|
||||
res = gst_rtspsrc_perform_seek (rtspsrc, event);
|
||||
gst_event_unref (event);
|
||||
} else {
|
||||
/* Store for later use */
|
||||
res = TRUE;
|
||||
rtspsrc->initial_seek = event;
|
||||
}
|
||||
} else if (GST_EVENT_IS_DOWNSTREAM (event)) {
|
||||
res = gst_rtspsrc_push_event (rtspsrc, event);
|
||||
} else {
|
||||
|
|
|
@ -277,6 +277,7 @@ struct _GstRTSPSrc {
|
|||
GstClockTime teardown_timeout;
|
||||
gboolean onvif_mode;
|
||||
gboolean onvif_rate_control;
|
||||
gboolean is_live;
|
||||
|
||||
/* state */
|
||||
GstRTSPState state;
|
||||
|
@ -320,6 +321,8 @@ struct _GstRTSPSrc {
|
|||
|
||||
GstRTSPVersion default_version;
|
||||
GstRTSPVersion version;
|
||||
|
||||
GstEvent *initial_seek;
|
||||
};
|
||||
|
||||
struct _GstRTSPSrcClass {
|
||||
|
|
Loading…
Reference in a new issue