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rtspsrc: expose and implement is-live property

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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:
Mathieu Duponchelle 2019-07-27 04:05:01 +02:00 committed by Mathieu Duponchelle
parent 75f53631e5
commit 5c7423d73c
2 changed files with 92 additions and 20 deletions
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109
gst/rtsp/gstrtspsrc.c
View file

@ -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 {

3
gst/rtsp/gstrtspsrc.h
View file

@ -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 {