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gstreamer/ext/webrtc/transportsendbin.c
Matthew Waters 1894293d63 webrtcbin: an element that handles the transport aspects of webrtc connections 
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/

The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer.  In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.

The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.

With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>

https://bugzilla.gnome.org/show_bug.cgi?id=792523
2018-02-02 15:02:21 +11:00

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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 "transportsendbin.h"
#include "utils.h"
/*
* ,------------------------transport_send_%u-------------------------,
* ; ,-----dtlssrtpenc---, ;
* rtp_sink o--------------------------o rtp_sink_0 ; ,---nicesink---, ;
* ; ; src o--o sink ; ;
* ; ,--outputselector--, ,-o rtcp_sink_0 ; '--------------' ;
* ; ; src_0 o-' '-------------------' ;
* rtcp_sink ;---o sink ; ,----dtlssrtpenc----, ,---nicesink---, ;
* ; ; src_1 o---o rtcp_sink_0 src o--o sink ; ;
* ; '------------------' '-------------------' '--------------' ;
* '------------------------------------------------------------------'
*
* outputselecter is used to switch between rtcp-mux and no rtcp-mux
*
* FIXME: Do we need a valve drop=TRUE for the no RTCP case?
*/
#define GST_CAT_DEFAULT gst_webrtc_transport_send_bin_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define transport_send_bin_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (TransportSendBin, transport_send_bin, GST_TYPE_BIN,
GST_DEBUG_CATEGORY_INIT (gst_webrtc_transport_send_bin_debug,
"webrtctransportsendbin", 0, "webrtctransportsendbin"););
static GstStaticPadTemplate rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtp_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStaticPadTemplate rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp"));
enum
{
PROP_0,
PROP_STREAM,
PROP_RTCP_MUX,
};
static void
_set_rtcp_mux (TransportSendBin * send, gboolean rtcp_mux)
{
GstPad *active_pad;
if (rtcp_mux)
active_pad = gst_element_get_static_pad (send->outputselector, "src_0");
else
active_pad = gst_element_get_static_pad (send->outputselector, "src_1");
send->rtcp_mux = rtcp_mux;
GST_OBJECT_UNLOCK (send);
g_object_set (send->outputselector, "active-pad", active_pad, NULL);
gst_object_unref (active_pad);
GST_OBJECT_LOCK (send);
}
static void
transport_send_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
TransportSendBin *send = TRANSPORT_SEND_BIN (object);
GST_OBJECT_LOCK (send);
switch (prop_id) {
case PROP_STREAM:
/* XXX: weak-ref this? */
send->stream = TRANSPORT_STREAM (g_value_get_object (value));
break;
case PROP_RTCP_MUX:
_set_rtcp_mux (send, g_value_get_boolean (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (send);
}
static void
transport_send_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
TransportSendBin *send = TRANSPORT_SEND_BIN (object);
GST_OBJECT_LOCK (send);
switch (prop_id) {
case PROP_STREAM:
g_value_set_object (value, send->stream);
break;
case PROP_RTCP_MUX:
g_value_set_boolean (value, send->rtcp_mux);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (send);
}
static GstPadProbeReturn
pad_block (GstPad * pad, GstPadProbeInfo * info, gpointer unused)
{
GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data);
return GST_PAD_PROBE_OK;
}
static GstStateChangeReturn
transport_send_bin_change_state (GstElement * element,
GstStateChange transition)
{
TransportSendBin *send = TRANSPORT_SEND_BIN (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GST_DEBUG_OBJECT (element, "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:{
/* XXX: don't change state until the client-ness has been chosen
* arguably the element should be able to deal with this itself or
* we should only add it once/if we get the encoding keys */
gst_element_set_locked_state (send->stream->transport->dtlssrtpenc, TRUE);
gst_element_set_locked_state (send->stream->rtcp_transport->dtlssrtpenc,
TRUE);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:{
GstElement *elem;
GstPad *pad;
/* unblock the encoder once the key is set, this should also be automatic */
elem = send->stream->transport->dtlssrtpenc;
pad = gst_element_get_static_pad (elem, "rtp_sink_0");
send->rtp_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtp_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
/* unblock the encoder once the key is set, this should also be automatic */
pad = gst_element_get_static_pad (elem, "rtcp_sink_0");
send->rtcp_mux_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtcp_mux_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
elem = send->stream->rtcp_transport->dtlssrtpenc;
/* unblock the encoder once the key is set, this should also be automatic */
pad = gst_element_get_static_pad (elem, "rtcp_sink_0");
send->rtcp_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtcp_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
/* unblock ice sink once a connection is made, this should also be automatic */
elem = send->stream->transport->transport->sink;
pad = gst_element_get_static_pad (elem, "sink");
send->rtp_nice_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtp_nice_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
/* unblock ice sink once a connection is made, this should also be automatic */
elem = send->stream->rtcp_transport->transport->sink;
pad = gst_element_get_static_pad (elem, "sink");
send->rtcp_nice_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtcp_nice_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
break;
}
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
/* Release pad blocks */
if (send->rtp_block && send->rtp_block->block_id) {
gst_pad_set_active (send->rtp_block->pad, FALSE);
gst_pad_remove_probe (send->rtp_block->pad, send->rtp_block->block_id);
send->rtp_block->block_id = 0;
}
if (send->rtcp_mux_block && send->rtcp_mux_block->block_id) {
gst_pad_set_active (send->rtcp_mux_block->pad, FALSE);
gst_pad_remove_probe (send->rtcp_mux_block->pad,
send->rtcp_mux_block->block_id);
send->rtcp_mux_block->block_id = 0;
}
if (send->rtcp_block && send->rtcp_block->block_id) {
gst_pad_set_active (send->rtcp_block->pad, FALSE);
gst_pad_remove_probe (send->rtcp_block->pad,
send->rtcp_block->block_id);
send->rtcp_block->block_id = 0;
}
if (send->rtp_nice_block && send->rtp_nice_block->block_id) {
gst_pad_set_active (send->rtp_nice_block->pad, FALSE);
gst_pad_remove_probe (send->rtp_nice_block->pad,
send->rtp_nice_block->block_id);
send->rtp_nice_block->block_id = 0;
}
if (send->rtcp_nice_block && send->rtcp_nice_block->block_id) {
gst_pad_set_active (send->rtcp_nice_block->pad, FALSE);
gst_pad_remove_probe (send->rtcp_nice_block->pad,
send->rtcp_nice_block->block_id);
send->rtcp_nice_block->block_id = 0;
}
break;
}
case GST_STATE_CHANGE_READY_TO_NULL:{
GstElement *elem;
if (send->rtp_block)
_free_pad_block (send->rtp_block);
send->rtp_block = NULL;
if (send->rtcp_mux_block)
_free_pad_block (send->rtcp_mux_block);
send->rtcp_mux_block = NULL;
elem = send->stream->transport->dtlssrtpenc;
gst_element_set_locked_state (elem, FALSE);
if (send->rtcp_block)
_free_pad_block (send->rtcp_block);
send->rtcp_block = NULL;
elem = send->stream->rtcp_transport->dtlssrtpenc;
gst_element_set_locked_state (elem, FALSE);
if (send->rtp_nice_block)
_free_pad_block (send->rtp_nice_block);
send->rtp_nice_block = NULL;
if (send->rtcp_nice_block)
_free_pad_block (send->rtcp_nice_block);
send->rtcp_nice_block = NULL;
break;
}
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return ret;
}
static void
_on_dtls_enc_key_set (GstElement * element, TransportSendBin * send)
{
if (element == send->stream->transport->dtlssrtpenc) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtp_block->pad);
_free_pad_block (send->rtp_block);
send->rtp_block = NULL;
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtcp_mux_block->pad);
_free_pad_block (send->rtcp_mux_block);
send->rtcp_mux_block = NULL;
} else if (element == send->stream->rtcp_transport->dtlssrtpenc) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtcp_block->pad);
_free_pad_block (send->rtcp_block);
send->rtcp_block = NULL;
}
}
static void
_on_notify_ice_connection_state (GstWebRTCICETransport * transport,
GParamSpec * pspec, TransportSendBin * send)
{
GstWebRTCICEConnectionState state;
g_object_get (transport, "state", &state, NULL);
if (state == GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED ||
state == GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED) {
GST_OBJECT_LOCK (send);
if (transport == send->stream->transport->transport) {
if (send->rtp_nice_block) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtp_nice_block->pad);
_free_pad_block (send->rtp_nice_block);
}
send->rtp_nice_block = NULL;
} else if (transport == send->stream->rtcp_transport->transport) {
if (send->rtcp_nice_block) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtcp_nice_block->pad);
_free_pad_block (send->rtcp_nice_block);
}
send->rtcp_nice_block = NULL;
}
GST_OBJECT_UNLOCK (send);
}
}
static void
transport_send_bin_constructed (GObject * object)
{
TransportSendBin *send = TRANSPORT_SEND_BIN (object);
GstWebRTCDTLSTransport *transport;
GstPadTemplate *templ;
GstPad *ghost, *pad;
g_return_if_fail (send->stream);
g_object_bind_property (send, "rtcp-mux", send->stream, "rtcp-mux",
G_BINDING_BIDIRECTIONAL);
transport = send->stream->transport;
templ = _find_pad_template (transport->dtlssrtpenc,
GST_PAD_SINK, GST_PAD_REQUEST, "rtp_sink_%d");
pad = gst_element_request_pad (transport->dtlssrtpenc, templ, "rtp_sink_0",
NULL);
/* unblock the encoder once the key is set */
g_signal_connect (transport->dtlssrtpenc, "on-key-set",
G_CALLBACK (_on_dtls_enc_key_set), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->dtlssrtpenc));
/* unblock ice sink once it signals a connection */
g_signal_connect (transport->transport, "notify::state",
G_CALLBACK (_on_notify_ice_connection_state), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->transport->sink));
if (!gst_element_link_pads (GST_ELEMENT (transport->dtlssrtpenc), "src",
GST_ELEMENT (transport->transport->sink), "sink"))
g_warn_if_reached ();
send->outputselector = gst_element_factory_make ("output-selector", NULL);
gst_bin_add (GST_BIN (send), send->outputselector);
if (!gst_element_link_pads (GST_ELEMENT (send->outputselector), "src_0",
GST_ELEMENT (transport->dtlssrtpenc), "rtcp_sink_0"))
g_warn_if_reached ();
ghost = gst_ghost_pad_new ("rtp_sink", pad);
gst_element_add_pad (GST_ELEMENT (send), ghost);
gst_object_unref (pad);
transport = send->stream->rtcp_transport;
templ = _find_pad_template (transport->dtlssrtpenc,
GST_PAD_SINK, GST_PAD_REQUEST, "rtcp_sink_%d");
/* unblock the encoder once the key is set */
g_signal_connect (transport->dtlssrtpenc, "on-key-set",
G_CALLBACK (_on_dtls_enc_key_set), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->dtlssrtpenc));
/* unblock ice sink once it signals a connection */
g_signal_connect (transport->transport, "notify::state",
G_CALLBACK (_on_notify_ice_connection_state), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->transport->sink));
if (!gst_element_link_pads (GST_ELEMENT (transport->dtlssrtpenc), "src",
GST_ELEMENT (transport->transport->sink), "sink"))
g_warn_if_reached ();
if (!gst_element_link_pads (GST_ELEMENT (send->outputselector), "src_1",
GST_ELEMENT (transport->dtlssrtpenc), "rtcp_sink_0"))
g_warn_if_reached ();
pad = gst_element_get_static_pad (send->outputselector, "sink");
ghost = gst_ghost_pad_new ("rtcp_sink", pad);
gst_element_add_pad (GST_ELEMENT (send), ghost);
gst_object_unref (pad);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
transport_send_bin_dispose (GObject * object)
{
TransportSendBin *send = TRANSPORT_SEND_BIN (object);
if (send->stream) {
g_signal_handlers_disconnect_by_data (send->stream->transport->transport,
send);
g_signal_handlers_disconnect_by_data (send->stream->
rtcp_transport->transport, send);
}
send->stream = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
transport_send_bin_class_init (TransportSendBinClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) klass;
element_class->change_state = transport_send_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_set_metadata (element_class, "WebRTC Transport Send Bin",
"Filter/Network/WebRTC", "A bin for webrtc connections",
"Matthew Waters <matthew@centricular.com>");
gobject_class->constructed = transport_send_bin_constructed;
gobject_class->dispose = transport_send_bin_dispose;
gobject_class->get_property = transport_send_bin_get_property;
gobject_class->set_property = transport_send_bin_set_property;
g_object_class_install_property (gobject_class,
PROP_STREAM,
g_param_spec_object ("stream", "Stream",
"The TransportStream for this sending bin",
transport_stream_get_type (),
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RTCP_MUX,
g_param_spec_boolean ("rtcp-mux", "RTCP Mux",
"Whether RTCP packets are muxed with RTP packets",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
transport_send_bin_init (TransportSendBin * send)
{
}
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