gstreamer/subprojects/gst-plugins-bad/ext/webrtc/transportstream.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 "transportstream.h"
#include "transportsendbin.h"
#include "transportreceivebin.h"
#include "gstwebrtcbin.h"
#include "utils.h"
#include "gst/webrtc/webrtc-priv.h"
#define GST_CAT_DEFAULT transport_stream_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define transport_stream_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (TransportStream, transport_stream, GST_TYPE_OBJECT,
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "webrtctransportstream", 0,
"webrtctransportstream"););
enum
{
PROP_0,
PROP_WEBRTC,
PROP_SESSION_ID,
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 media_idx)
{
guint i;
gint ret = -1;
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
if (media_idx != -1 && media_idx != item->media_idx)
continue;
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_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_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);
gst_clear_object (&stream->send_bin);
gst_clear_object (&stream->receive_bin);
gst_clear_object (&stream->transport);
gst_clear_object (&stream->rtxsend);
gst_clear_object (&stream->rtxreceive);
gst_clear_object (&stream->reddec);
g_list_free_full (stream->fecdecs, (GDestroyNotify) gst_object_unref);
stream->fecdecs = 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_ptr_array_free (stream->ssrcmap, TRUE);
gst_clear_object (&stream->rtxsend_stream_id);
gst_clear_object (&stream->rtxsend_repaired_stream_id);
gst_clear_object (&stream->rtxreceive_stream_id);
gst_clear_object (&stream->rtxreceive_repaired_stream_id);
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);
webrtc = GST_WEBRTC_BIN (gst_object_get_parent (GST_OBJECT (object)));
g_object_bind_property (stream->transport, "client", stream, "dtls-client",
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);
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_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 SsrcMapItem *
ssrcmap_item_new (GstWebRTCRTPTransceiverDirection direction, guint32 ssrc,
guint media_idx)
{
SsrcMapItem *ssrc_item = g_new0 (SsrcMapItem, 1);
ssrc_item->direction = direction;
ssrc_item->media_idx = media_idx;
ssrc_item->ssrc = ssrc;
g_weak_ref_init (&ssrc_item->rtpjitterbuffer, NULL);
return ssrc_item;
}
static void
ssrcmap_item_free (SsrcMapItem * item)
{
g_weak_ref_clear (&item->rtpjitterbuffer);
g_clear_pointer (&item->mid, g_free);
g_clear_pointer (&item->rid, g_free);
g_free (item);
}
SsrcMapItem *
transport_stream_find_ssrc_map_item (TransportStream * stream,
gconstpointer data, FindSsrcMapFunc func)
{
int i;
for (i = 0; i < stream->ssrcmap->len; i++) {
SsrcMapItem *item = g_ptr_array_index (stream->ssrcmap, i);
if (func (item, data))
return item;
}
return NULL;
}
void
transport_stream_filter_ssrc_map_item (TransportStream * stream,
gconstpointer data, FindSsrcMapFunc func)
{
int i;
for (i = 0; i < stream->ssrcmap->len;) {
SsrcMapItem *item = g_ptr_array_index (stream->ssrcmap, i);
if (!func (item, data)) {
GST_TRACE_OBJECT (stream, "removing ssrc %u", item->ssrc);
g_ptr_array_remove_index_fast (stream->ssrcmap, i);
} else {
i++;
}
}
}
SsrcMapItem *
transport_stream_add_ssrc_map_item (TransportStream * stream,
GstWebRTCRTPTransceiverDirection direction, guint32 ssrc, guint media_idx)
{
SsrcMapItem *ret = NULL;
char *dir_str = gst_webrtc_rtp_transceiver_direction_to_string (direction);
g_return_val_if_fail (direction ==
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY
|| direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, NULL);
g_return_val_if_fail (ssrc != 0, NULL);
GST_INFO_OBJECT (stream, "Adding mapping for rtp session %u media_idx %u "
"direction %s ssrc %u", stream->session_id, media_idx, dir_str, ssrc);
/* XXX: duplicates? */
ret = ssrcmap_item_new (direction, ssrc, media_idx);
g_ptr_array_add (stream->ssrcmap, ret);
g_free (dir_str);
return ret;
}
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->ssrcmap = g_ptr_array_new_with_free_func (
(GDestroyNotify) ssrcmap_item_free);
stream->rtphdrext_id_stream_id = -1;
stream->rtphdrext_id_repaired_stream_id = -1;
}
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;
}