gstreamer/ext/webrtc/transportstream.c
Matthew Waters 177aa22bcd webrtc: Initial support for stream addition/removal
Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
  will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
2019-05-30 21:33:09 +10:00

327 lines
9.3 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 "transportstream.h"
#include "transportsendbin.h"
#include "transportreceivebin.h"
#include "gstwebrtcice.h"
#include "gstwebrtcbin.h"
#include "utils.h"
#define transport_stream_parent_class parent_class
G_DEFINE_TYPE (TransportStream, transport_stream, GST_TYPE_OBJECT);
enum
{
PROP_0,
PROP_WEBRTC,
PROP_SESSION_ID,
PROP_RTCP_MUX,
PROP_DTLS_CLIENT,
};
GstCaps *
transport_stream_get_caps_for_pt (TransportStream * stream, guint pt)
{
guint i, len;
len = stream->ptmap->len;
for (i = 0; i < len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (item->pt == pt)
return item->caps;
}
return NULL;
}
int
transport_stream_get_pt (TransportStream * stream, const gchar * encoding_name)
{
guint i;
gint ret = 0;
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (!gst_caps_is_empty (item->caps)) {
GstStructure *s = gst_caps_get_structure (item->caps, 0);
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
encoding_name)) {
ret = item->pt;
break;
}
}
}
return ret;
}
int *
transport_stream_get_all_pt (TransportStream * stream,
const gchar * encoding_name, gsize * pt_len)
{
guint i;
gsize ret_i = 0;
gsize ret_size = 8;
int *ret = NULL;
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (!gst_caps_is_empty (item->caps)) {
GstStructure *s = gst_caps_get_structure (item->caps, 0);
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"),
encoding_name)) {
if (!ret)
ret = g_new0 (int, ret_size);
if (ret_i >= ret_size) {
ret_size *= 2;
ret = g_realloc_n (ret, ret_size, sizeof (int));
}
ret[ret_i++] = item->pt;
}
}
}
*pt_len = ret_i;
return ret;
}
static void
transport_stream_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
TransportStream *stream = TRANSPORT_STREAM (object);
switch (prop_id) {
case PROP_WEBRTC:
gst_object_set_parent (GST_OBJECT (stream), g_value_get_object (value));
break;
}
GST_OBJECT_LOCK (stream);
switch (prop_id) {
case PROP_WEBRTC:
break;
case PROP_SESSION_ID:
stream->session_id = g_value_get_uint (value);
break;
case PROP_RTCP_MUX:
stream->rtcp_mux = g_value_get_boolean (value);
break;
case PROP_DTLS_CLIENT:
stream->dtls_client = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (stream);
}
static void
transport_stream_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
TransportStream *stream = TRANSPORT_STREAM (object);
GST_OBJECT_LOCK (stream);
switch (prop_id) {
case PROP_SESSION_ID:
g_value_set_uint (value, stream->session_id);
break;
case PROP_RTCP_MUX:
g_value_set_boolean (value, stream->rtcp_mux);
break;
case PROP_DTLS_CLIENT:
g_value_set_boolean (value, stream->dtls_client);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (stream);
}
static void
transport_stream_dispose (GObject * object)
{
TransportStream *stream = TRANSPORT_STREAM (object);
if (stream->send_bin)
gst_object_unref (stream->send_bin);
stream->send_bin = NULL;
if (stream->receive_bin)
gst_object_unref (stream->receive_bin);
stream->receive_bin = NULL;
if (stream->transport)
gst_object_unref (stream->transport);
stream->transport = NULL;
if (stream->rtcp_transport)
gst_object_unref (stream->rtcp_transport);
stream->rtcp_transport = NULL;
if (stream->rtxsend)
gst_object_unref (stream->rtxsend);
stream->rtxsend = NULL;
if (stream->rtxreceive)
gst_object_unref (stream->rtxreceive);
stream->rtxreceive = NULL;
GST_OBJECT_PARENT (object) = NULL;
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
transport_stream_finalize (GObject * object)
{
TransportStream *stream = TRANSPORT_STREAM (object);
g_array_free (stream->ptmap, TRUE);
g_array_free (stream->remote_ssrcmap, TRUE);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
transport_stream_constructed (GObject * object)
{
TransportStream *stream = TRANSPORT_STREAM (object);
GstWebRTCBin *webrtc;
GstWebRTCICETransport *ice_trans;
stream->transport = gst_webrtc_dtls_transport_new (stream->session_id, FALSE);
stream->rtcp_transport =
gst_webrtc_dtls_transport_new (stream->session_id, TRUE);
webrtc = GST_WEBRTC_BIN (gst_object_get_parent (GST_OBJECT (object)));
g_object_bind_property (stream->transport, "client", stream, "dtls-client",
G_BINDING_BIDIRECTIONAL);
g_object_bind_property (stream->rtcp_transport, "client", stream,
"dtls-client", G_BINDING_BIDIRECTIONAL);
g_object_bind_property (stream->transport, "certificate",
stream->rtcp_transport, "certificate", G_BINDING_BIDIRECTIONAL);
/* Need to go full Java and have a transport manager?
* Or make the caller set the ICE transport up? */
stream->stream = _find_ice_stream_for_session (webrtc, stream->session_id);
if (stream->stream == NULL) {
stream->stream = gst_webrtc_ice_add_stream (webrtc->priv->ice,
stream->session_id);
_add_ice_stream_item (webrtc, stream->session_id, stream->stream);
}
ice_trans =
gst_webrtc_ice_find_transport (webrtc->priv->ice, stream->stream,
GST_WEBRTC_ICE_COMPONENT_RTP);
gst_webrtc_dtls_transport_set_transport (stream->transport, ice_trans);
gst_object_unref (ice_trans);
ice_trans =
gst_webrtc_ice_find_transport (webrtc->priv->ice, stream->stream,
GST_WEBRTC_ICE_COMPONENT_RTCP);
gst_webrtc_dtls_transport_set_transport (stream->rtcp_transport, ice_trans);
gst_object_unref (ice_trans);
stream->send_bin = g_object_new (transport_send_bin_get_type (), "stream",
stream, NULL);
gst_object_ref_sink (stream->send_bin);
stream->receive_bin = g_object_new (transport_receive_bin_get_type (),
"stream", stream, NULL);
gst_object_ref_sink (stream->receive_bin);
gst_object_unref (webrtc);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
transport_stream_class_init (TransportStreamClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->constructed = transport_stream_constructed;
gobject_class->get_property = transport_stream_get_property;
gobject_class->set_property = transport_stream_set_property;
gobject_class->dispose = transport_stream_dispose;
gobject_class->finalize = transport_stream_finalize;
/* some acrobatics are required to set the parent before _constructed()
* has been called */
g_object_class_install_property (gobject_class,
PROP_WEBRTC,
g_param_spec_object ("webrtc", "Parent webrtcbin",
"Parent webrtcbin",
GST_TYPE_WEBRTC_BIN,
G_PARAM_WRITABLE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_SESSION_ID,
g_param_spec_uint ("session-id", "Session ID",
"Session ID used for this transport",
0, G_MAXUINT, 0,
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));
g_object_class_install_property (gobject_class,
PROP_DTLS_CLIENT,
g_param_spec_boolean ("dtls-client", "DTLS client",
"Whether we take the client role in DTLS negotiation",
FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
clear_ptmap_item (PtMapItem * item)
{
if (item->caps)
gst_caps_unref (item->caps);
}
static void
transport_stream_init (TransportStream * stream)
{
stream->ptmap = g_array_new (FALSE, TRUE, sizeof (PtMapItem));
g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
stream->remote_ssrcmap = g_array_new (FALSE, TRUE, sizeof (SsrcMapItem));
}
TransportStream *
transport_stream_new (GstWebRTCBin * webrtc, guint session_id)
{
TransportStream *stream;
stream = g_object_new (transport_stream_get_type (), "webrtc", webrtc,
"session-id", session_id, NULL);
return stream;
}