gstreamer/ext/webrtc/transportreceivebin.c

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/* 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_DROP:
return "drop";
case RECEIVE_STATE_PASS:
return "pass";
default:
return "Unknown";
}
}
static GstPadProbeReturn
pad_block (GstPad * pad, GstPadProbeInfo * info, TransportReceiveBin * receive)
{
g_mutex_lock (&receive->pad_block_lock);
while (receive->receive_state == RECEIVE_STATE_BLOCK) {
g_cond_wait (&receive->pad_block_cond, &receive->pad_block_lock);
GST_DEBUG_OBJECT (pad, "probe waited. new state %s",
_receive_state_to_string (receive->receive_state));
}
g_mutex_unlock (&receive->pad_block_lock);
return GST_PAD_PROBE_OK;
}
static GstPadProbeReturn
src_probe_cb (GstPad * pad, GstPadProbeInfo * info,
TransportReceiveBin * receive)
{
GstPadProbeReturn ret;
g_mutex_lock (&receive->pad_block_lock);
g_assert (receive->receive_state != RECEIVE_STATE_BLOCK);
ret =
receive->receive_state ==
RECEIVE_STATE_DROP ? GST_PAD_PROBE_DROP : GST_PAD_PROBE_OK;
g_mutex_unlock (&receive->pad_block_lock);
return ret;
}
void
transport_receive_bin_set_receive_state (TransportReceiveBin * receive,
ReceiveState state)
{
g_mutex_lock (&receive->pad_block_lock);
receive->receive_state = state;
GST_DEBUG_OBJECT (receive, "changing receive state to %s",
_receive_state_to_string (state));
g_cond_signal (&receive->pad_block_cond);
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_cond_clear (&receive->pad_block_cond);
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:{
GstWebRTCDTLSTransport *transport;
GstElement *elem, *dtlssrtpdec;
GstPad *pad;
transport = receive->stream->transport;
dtlssrtpdec = transport->dtlssrtpdec;
pad = gst_element_get_static_pad (dtlssrtpdec, "sink");
receive->rtp_block =
_create_pad_block (GST_ELEMENT (receive), pad, 0, NULL, NULL);
receive->rtp_block->block_id =
gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_ALL_BOTH,
(GstPadProbeCallback) pad_block, receive, NULL);
gst_object_unref (pad);
receive->rtp_src_probe_id = gst_pad_add_probe (receive->rtp_src,
GST_PAD_PROBE_TYPE_ALL_BOTH, (GstPadProbeCallback) src_probe_cb,
receive, NULL);
transport = receive->stream->rtcp_transport;
dtlssrtpdec = transport->dtlssrtpdec;
pad = gst_element_get_static_pad (dtlssrtpdec, "sink");
receive->rtcp_block =
_create_pad_block (GST_ELEMENT (receive), pad, 0, NULL, NULL);
receive->rtcp_block->block_id =
gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_ALL_BOTH,
(GstPadProbeCallback) pad_block, receive, NULL);
gst_object_unref (pad);
receive->rtcp_src_probe_id = gst_pad_add_probe (receive->rtcp_src,
GST_PAD_PROBE_TYPE_ALL_BOTH, (GstPadProbeCallback) src_probe_cb,
receive, NULL);
/* 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->rtp_src_probe_id)
gst_pad_remove_probe (receive->rtp_src, receive->rtp_src_probe_id);
receive->rtp_src_probe_id = 0;
if (receive->rtcp_block)
_free_pad_block (receive->rtcp_block);
receive->rtcp_block = NULL;
if (receive->rtcp_src_probe_id)
gst_pad_remove_probe (receive->rtcp_src, receive->rtcp_src_probe_id);
receive->rtcp_src_probe_id = 0;
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",
2019-09-02 19:08:44 +00:00
"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)
{
g_mutex_init (&receive->pad_block_lock);
g_cond_init (&receive->pad_block_cond);
}