mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-08 18:39:54 +00:00
cb48733ff3
As per discussion in the bug, remove the drop state from transportreceivebin. Dropping data is necessary, but for bundled config, needs to happen further downstream after mixed flows have been separated. Also support switching back to BLOCK from PASS state. https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1206
452 lines
16 KiB
C
452 lines
16 KiB
C
/* GStreamer
|
|
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Library General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2 of the License, or (at your option) any later version.
|
|
*
|
|
* This library is distributed in the hope that it will be useful,
|
|
* but WITHOUT ANY WARRANTY; without even the implied warranty of
|
|
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
|
|
* Library General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Library General Public
|
|
* License along with this library; if not, write to the
|
|
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, USA.
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
# include "config.h"
|
|
#endif
|
|
|
|
#include "transportreceivebin.h"
|
|
#include "utils.h"
|
|
|
|
/*
|
|
* ,----------------------------transport_receive_%u---------------------------,
|
|
* ; (rtp/data) ;
|
|
* ; ,-nicesrc-, ,-capsfilter-, ,--queue--, ,-dtlssrtpdec-, ,-funnel-, ;
|
|
* ; ; src o-o sink src o-osink srco-osink rtp_srco-------o sink_0 ; ;
|
|
* ; '---------' '------------' '---------' ; ; ; src o--o rtp_src
|
|
* ; ; rtcp_srco---, ,-o sink_1 ; ;
|
|
* ; ; ; ; ; '--------' ;
|
|
* ; ; data_srco-, ; ; ,-funnel-, ;
|
|
* ; (rtcp) '-------------' ; '-+-o sink_0 ; ;
|
|
* ; ,-nicesrc-, ,-capsfilter-, ,--queue--, ,-dtlssrtpdec-, ; ,-' ; src o--o rtcp_src
|
|
* ; ; src o-o sink src o-osink srco-osink rtp_srco-+-' ,-o sink_1 ; ;
|
|
* ; '---------' '------------' '---------' ; ; ; ; '--------' ;
|
|
* ; ; rtcp_srco-+---' ,-funnel-, ;
|
|
* ; ; ; '-----o sink_0 ; ;
|
|
* ; ; data_srco-, ; src o--o data_src
|
|
* ; '-------------' '-----o sink_1 ; ;
|
|
* ; '--------' ;
|
|
* '---------------------------------------------------------------------------'
|
|
*
|
|
* Do we really wnat to be *that* permissive in what we accept?
|
|
*
|
|
* FIXME: When and how do we want to clear the possibly stored buffers?
|
|
*/
|
|
|
|
#define GST_CAT_DEFAULT gst_webrtc_transport_receive_bin_debug
|
|
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
|
|
|
|
#define transport_receive_bin_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_CODE (TransportReceiveBin, transport_receive_bin,
|
|
GST_TYPE_BIN,
|
|
GST_DEBUG_CATEGORY_INIT (gst_webrtc_transport_receive_bin_debug,
|
|
"webrtctransportreceivebin", 0, "webrtctransportreceivebin");
|
|
);
|
|
|
|
static GstStaticPadTemplate rtp_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("rtp_src",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("application/x-rtp"));
|
|
|
|
static GstStaticPadTemplate rtcp_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("rtcp_src",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("application/x-rtp"));
|
|
|
|
static GstStaticPadTemplate data_sink_template =
|
|
GST_STATIC_PAD_TEMPLATE ("data_src",
|
|
GST_PAD_SINK,
|
|
GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS_ANY);
|
|
|
|
enum
|
|
{
|
|
PROP_0,
|
|
PROP_STREAM,
|
|
};
|
|
|
|
static const gchar *
|
|
_receive_state_to_string (ReceiveState state)
|
|
{
|
|
switch (state) {
|
|
case RECEIVE_STATE_BLOCK:
|
|
return "block";
|
|
case RECEIVE_STATE_PASS:
|
|
return "pass";
|
|
default:
|
|
return "Unknown";
|
|
}
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
pad_block (GstPad * pad, GstPadProbeInfo * info, TransportReceiveBin * receive)
|
|
{
|
|
/* Drop all events: we don't care about them and don't want to block on
|
|
* them. Sticky events would be forwarded again later once we unblock
|
|
* and we don't want to forward them here already because that might
|
|
* cause a spurious GST_FLOW_FLUSHING */
|
|
if (GST_IS_EVENT (info->data))
|
|
return GST_PAD_PROBE_DROP;
|
|
|
|
/* But block on any actual data-flow so we don't accidentally send that
|
|
* to a pad that is not ready yet, causing GST_FLOW_FLUSHING and everything
|
|
* to silently stop.
|
|
*/
|
|
GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data);
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
void
|
|
transport_receive_bin_set_receive_state (TransportReceiveBin * receive,
|
|
ReceiveState state)
|
|
{
|
|
|
|
g_mutex_lock (&receive->pad_block_lock);
|
|
if (receive->receive_state != state) {
|
|
GST_DEBUG_OBJECT (receive, "changing receive state to %s",
|
|
_receive_state_to_string (state));
|
|
}
|
|
|
|
if (state == RECEIVE_STATE_PASS) {
|
|
if (receive->rtp_block)
|
|
_free_pad_block (receive->rtp_block);
|
|
receive->rtp_block = NULL;
|
|
|
|
if (receive->rtcp_block)
|
|
_free_pad_block (receive->rtcp_block);
|
|
receive->rtcp_block = NULL;
|
|
} else {
|
|
g_assert (state == RECEIVE_STATE_BLOCK);
|
|
if (receive->rtp_block == NULL) {
|
|
GstWebRTCDTLSTransport *transport;
|
|
GstElement *dtlssrtpdec;
|
|
GstPad *pad, *peer_pad;
|
|
|
|
if (receive->stream) {
|
|
transport = receive->stream->transport;
|
|
dtlssrtpdec = transport->dtlssrtpdec;
|
|
pad = gst_element_get_static_pad (dtlssrtpdec, "sink");
|
|
peer_pad = gst_pad_get_peer (pad);
|
|
receive->rtp_block =
|
|
_create_pad_block (GST_ELEMENT (receive), peer_pad, 0, NULL, NULL);
|
|
receive->rtp_block->block_id =
|
|
gst_pad_add_probe (peer_pad,
|
|
GST_PAD_PROBE_TYPE_BLOCK |
|
|
GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM,
|
|
(GstPadProbeCallback) pad_block, receive, NULL);
|
|
gst_object_unref (peer_pad);
|
|
gst_object_unref (pad);
|
|
|
|
transport = receive->stream->rtcp_transport;
|
|
dtlssrtpdec = transport->dtlssrtpdec;
|
|
pad = gst_element_get_static_pad (dtlssrtpdec, "sink");
|
|
peer_pad = gst_pad_get_peer (pad);
|
|
receive->rtcp_block =
|
|
_create_pad_block (GST_ELEMENT (receive), peer_pad, 0, NULL, NULL);
|
|
receive->rtcp_block->block_id =
|
|
gst_pad_add_probe (peer_pad,
|
|
GST_PAD_PROBE_TYPE_BLOCK |
|
|
GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM,
|
|
(GstPadProbeCallback) pad_block, receive, NULL);
|
|
gst_object_unref (peer_pad);
|
|
gst_object_unref (pad);
|
|
}
|
|
}
|
|
}
|
|
|
|
receive->receive_state = state;
|
|
g_mutex_unlock (&receive->pad_block_lock);
|
|
}
|
|
|
|
static void
|
|
transport_receive_bin_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
|
|
|
|
GST_OBJECT_LOCK (receive);
|
|
switch (prop_id) {
|
|
case PROP_STREAM:
|
|
/* XXX: weak-ref this? */
|
|
receive->stream = TRANSPORT_STREAM (g_value_get_object (value));
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
GST_OBJECT_UNLOCK (receive);
|
|
}
|
|
|
|
static void
|
|
transport_receive_bin_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
|
|
|
|
GST_OBJECT_LOCK (receive);
|
|
switch (prop_id) {
|
|
case PROP_STREAM:
|
|
g_value_set_object (value, receive->stream);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
GST_OBJECT_UNLOCK (receive);
|
|
}
|
|
|
|
static void
|
|
transport_receive_bin_finalize (GObject * object)
|
|
{
|
|
TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
|
|
|
|
g_mutex_clear (&receive->pad_block_lock);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
transport_receive_bin_change_state (GstElement * element,
|
|
GstStateChange transition)
|
|
{
|
|
TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (element);
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
|
|
GST_DEBUG ("changing state: %s => %s",
|
|
gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
|
|
gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:{
|
|
GstElement *elem;
|
|
|
|
/* We want to start blocked, unless someone already switched us
|
|
* to PASS mode. receive_state is set to BLOCKED in _init(),
|
|
* so set up blocks with whatever the mode is now. */
|
|
transport_receive_bin_set_receive_state (receive, receive->receive_state);
|
|
|
|
/* XXX: because nice needs the nicesrc internal main loop running in order
|
|
* correctly STUN... */
|
|
/* FIXME: this races with the pad exposure later and may get not-linked */
|
|
elem = receive->stream->transport->transport->src;
|
|
gst_element_set_locked_state (elem, TRUE);
|
|
gst_element_set_state (elem, GST_STATE_PLAYING);
|
|
elem = receive->stream->rtcp_transport->transport->src;
|
|
gst_element_set_locked_state (elem, TRUE);
|
|
gst_element_set_state (elem, GST_STATE_PLAYING);
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
return ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_READY_TO_NULL:{
|
|
GstElement *elem;
|
|
|
|
elem = receive->stream->transport->transport->src;
|
|
gst_element_set_locked_state (elem, FALSE);
|
|
gst_element_set_state (elem, GST_STATE_NULL);
|
|
elem = receive->stream->rtcp_transport->transport->src;
|
|
gst_element_set_locked_state (elem, FALSE);
|
|
gst_element_set_state (elem, GST_STATE_NULL);
|
|
|
|
if (receive->rtp_block)
|
|
_free_pad_block (receive->rtp_block);
|
|
receive->rtp_block = NULL;
|
|
|
|
if (receive->rtcp_block)
|
|
_free_pad_block (receive->rtcp_block);
|
|
receive->rtcp_block = NULL;
|
|
|
|
break;
|
|
}
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
rtp_queue_overrun (GstElement * queue, TransportReceiveBin * receive)
|
|
{
|
|
GST_WARNING_OBJECT (receive, "Internal receive queue overrun. Dropping data");
|
|
}
|
|
|
|
static void
|
|
transport_receive_bin_constructed (GObject * object)
|
|
{
|
|
TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
|
|
GstWebRTCDTLSTransport *transport;
|
|
GstPad *ghost, *pad;
|
|
GstElement *capsfilter, *funnel, *queue;
|
|
GstCaps *caps;
|
|
|
|
g_return_if_fail (receive->stream);
|
|
|
|
/* link ice src, dtlsrtp together for rtp */
|
|
transport = receive->stream->transport;
|
|
gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->dtlssrtpdec));
|
|
|
|
capsfilter = gst_element_factory_make ("capsfilter", NULL);
|
|
caps = gst_caps_new_empty_simple ("application/x-rtp");
|
|
g_object_set (capsfilter, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
queue = gst_element_factory_make ("queue", NULL);
|
|
/* FIXME: make this configurable? */
|
|
g_object_set (queue, "leaky", 2, "max-size-time", (guint64) 0,
|
|
"max-size-buffers", 0, "max-size-bytes", 5 * 1024 * 1024, NULL);
|
|
g_signal_connect (queue, "overrun", G_CALLBACK (rtp_queue_overrun), receive);
|
|
|
|
gst_bin_add (GST_BIN (receive), GST_ELEMENT (queue));
|
|
gst_bin_add (GST_BIN (receive), GST_ELEMENT (capsfilter));
|
|
if (!gst_element_link_pads (capsfilter, "src", queue, "sink"))
|
|
g_warn_if_reached ();
|
|
|
|
if (!gst_element_link_pads (queue, "src", transport->dtlssrtpdec, "sink"))
|
|
g_warn_if_reached ();
|
|
|
|
gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->transport->src));
|
|
if (!gst_element_link_pads (GST_ELEMENT (transport->transport->src), "src",
|
|
GST_ELEMENT (capsfilter), "sink"))
|
|
g_warn_if_reached ();
|
|
|
|
/* link ice src, dtlsrtp together for rtcp */
|
|
transport = receive->stream->rtcp_transport;
|
|
gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->dtlssrtpdec));
|
|
|
|
capsfilter = gst_element_factory_make ("capsfilter", NULL);
|
|
caps = gst_caps_new_empty_simple ("application/x-rtcp");
|
|
g_object_set (capsfilter, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
queue = gst_element_factory_make ("queue", NULL);
|
|
/* FIXME: make this configurable? */
|
|
g_object_set (queue, "leaky", 2, "max-size-time", (guint64) 0,
|
|
"max-size-buffers", 0, "max-size-bytes", 5 * 1024 * 1024, NULL);
|
|
g_signal_connect (queue, "overrun", G_CALLBACK (rtp_queue_overrun), receive);
|
|
|
|
gst_bin_add (GST_BIN (receive), queue);
|
|
gst_bin_add (GST_BIN (receive), GST_ELEMENT (capsfilter));
|
|
if (!gst_element_link_pads (capsfilter, "src", queue, "sink"))
|
|
g_warn_if_reached ();
|
|
|
|
if (!gst_element_link_pads (queue, "src", transport->dtlssrtpdec, "sink"))
|
|
g_warn_if_reached ();
|
|
|
|
gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->transport->src));
|
|
if (!gst_element_link_pads (GST_ELEMENT (transport->transport->src), "src",
|
|
GST_ELEMENT (capsfilter), "sink"))
|
|
g_warn_if_reached ();
|
|
|
|
/* create funnel for rtp_src */
|
|
funnel = gst_element_factory_make ("funnel", NULL);
|
|
gst_bin_add (GST_BIN (receive), funnel);
|
|
if (!gst_element_link_pads (receive->stream->transport->dtlssrtpdec,
|
|
"rtp_src", funnel, "sink_0"))
|
|
g_warn_if_reached ();
|
|
if (!gst_element_link_pads (receive->stream->rtcp_transport->dtlssrtpdec,
|
|
"rtp_src", funnel, "sink_1"))
|
|
g_warn_if_reached ();
|
|
|
|
pad = gst_element_get_static_pad (funnel, "src");
|
|
receive->rtp_src = gst_ghost_pad_new ("rtp_src", pad);
|
|
|
|
gst_element_add_pad (GST_ELEMENT (receive), receive->rtp_src);
|
|
gst_object_unref (pad);
|
|
|
|
/* create funnel for rtcp_src */
|
|
funnel = gst_element_factory_make ("funnel", NULL);
|
|
gst_bin_add (GST_BIN (receive), funnel);
|
|
if (!gst_element_link_pads (receive->stream->transport->dtlssrtpdec,
|
|
"rtcp_src", funnel, "sink_0"))
|
|
g_warn_if_reached ();
|
|
if (!gst_element_link_pads (receive->stream->rtcp_transport->dtlssrtpdec,
|
|
"rtcp_src", funnel, "sink_1"))
|
|
g_warn_if_reached ();
|
|
|
|
pad = gst_element_get_static_pad (funnel, "src");
|
|
receive->rtcp_src = gst_ghost_pad_new ("rtcp_src", pad);
|
|
gst_element_add_pad (GST_ELEMENT (receive), receive->rtcp_src);
|
|
gst_object_unref (pad);
|
|
|
|
/* create funnel for data_src */
|
|
funnel = gst_element_factory_make ("funnel", NULL);
|
|
gst_bin_add (GST_BIN (receive), funnel);
|
|
if (!gst_element_link_pads (receive->stream->transport->dtlssrtpdec,
|
|
"data_src", funnel, "sink_0"))
|
|
g_warn_if_reached ();
|
|
if (!gst_element_link_pads (receive->stream->rtcp_transport->dtlssrtpdec,
|
|
"data_src", funnel, "sink_1"))
|
|
g_warn_if_reached ();
|
|
|
|
pad = gst_element_get_static_pad (funnel, "src");
|
|
ghost = gst_ghost_pad_new ("data_src", pad);
|
|
gst_element_add_pad (GST_ELEMENT (receive), ghost);
|
|
gst_object_unref (pad);
|
|
|
|
G_OBJECT_CLASS (parent_class)->constructed (object);
|
|
}
|
|
|
|
static void
|
|
transport_receive_bin_class_init (TransportReceiveBinClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = (GObjectClass *) klass;
|
|
GstElementClass *element_class = (GstElementClass *) klass;
|
|
|
|
element_class->change_state = transport_receive_bin_change_state;
|
|
|
|
gst_element_class_add_static_pad_template (element_class, &rtp_sink_template);
|
|
gst_element_class_add_static_pad_template (element_class,
|
|
&rtcp_sink_template);
|
|
gst_element_class_add_static_pad_template (element_class,
|
|
&data_sink_template);
|
|
|
|
gst_element_class_set_metadata (element_class, "WebRTC Transport Receive Bin",
|
|
"Filter/Network/WebRTC", "A bin for webrtc connections",
|
|
"Matthew Waters <matthew@centricular.com>");
|
|
|
|
gobject_class->constructed = transport_receive_bin_constructed;
|
|
gobject_class->get_property = transport_receive_bin_get_property;
|
|
gobject_class->set_property = transport_receive_bin_set_property;
|
|
gobject_class->finalize = transport_receive_bin_finalize;
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_STREAM,
|
|
g_param_spec_object ("stream", "Stream",
|
|
"The TransportStream for this receiving bin",
|
|
transport_stream_get_type (),
|
|
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
|
|
}
|
|
|
|
static void
|
|
transport_receive_bin_init (TransportReceiveBin * receive)
|
|
{
|
|
receive->receive_state = RECEIVE_STATE_BLOCK;
|
|
g_mutex_init (&receive->pad_block_lock);
|
|
}
|