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webrtc: Remove non rtcp-mux code

Browse source 
RTCP mux is now always required by the WebRTC spec

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
This commit is contained in:
Olivier Crête 2020-11-02 19:49:55 -05:00 committed by GStreamer Merge Bot
parent 64f96facca
commit 5d5417f271
14 changed files with 46 additions and 379 deletions
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102
ext/webrtc/gstwebrtcbin.c
View file

@ -80,10 +80,10 @@
* configuration. Some cases are outlined below for a simple single
* audio/video/data session:
*
* - max-bundle (requires rtcp-muxing) uses a single transport for all
* - max-bundle uses a single transport for all
* media/data transported. Renegotiation involves adding/removing the
* necessary streams to the existing transports.
* - max-compat without rtcp-mux involves two TransportStream per media stream
* - max-compat involves two TransportStream per media stream
* to transport the rtp and the rtcp packets and a single TransportStream for
* all data channels. Each stream change involves modifying the associated
* TransportStream/s as necessary.
@ -859,11 +859,8 @@ _collate_ice_connection_states (GstWebRTCBin * webrtc)
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *rtp_trans =
g_ptr_array_index (webrtc->priv->transceivers, i);
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
TransportStream *stream = trans->stream;
GstWebRTCICETransport *transport, *rtcp_transport;
GstWebRTCICETransport *transport;
GstWebRTCICEConnectionState ice_state;
gboolean rtcp_mux = FALSE;
if (rtp_trans->stopped) {
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
@ -875,8 +872,6 @@ _collate_ice_connection_states (GstWebRTCBin * webrtc)
continue;
}
g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL);
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
/* get transport state */
@ -892,24 +887,6 @@ _collate_ice_connection_states (GstWebRTCBin * webrtc)
if (ice_state != STATE (CONNECTED) && ice_state != STATE (COMPLETED)
&& ice_state != STATE (CLOSED))
all_connected_completed_or_closed = FALSE;
rtcp_transport =
webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans)->transport;
if (!rtcp_mux && rtcp_transport && transport != rtcp_transport) {
g_object_get (rtcp_transport, "state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p RTCP state 0x%x", rtp_trans,
ice_state);
any_state |= (1 << ice_state);
if (ice_state != STATE (NEW) && ice_state != STATE (CLOSED))
all_new_or_closed = FALSE;
if (ice_state != STATE (COMPLETED) && ice_state != STATE (CLOSED))
all_completed_or_closed = FALSE;
if (ice_state != STATE (CONNECTED) && ice_state != STATE (COMPLETED)
&& ice_state != STATE (CLOSED))
all_connected_completed_or_closed = FALSE;
}
}
GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x", any_state);
@ -970,7 +947,7 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
TransportStream *stream = trans->stream;
GstWebRTCDTLSTransport *dtls_transport;
GstWebRTCICETransport *transport, *rtcp_transport;
GstWebRTCICETransport *transport;
GstWebRTCICEGatheringState ice_state;
gboolean rtcp_mux = FALSE;
@ -1003,22 +980,6 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc)
any_state |= (1 << ice_state);
if (ice_state != STATE (COMPLETE))
all_completed = FALSE;
dtls_transport = webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans);
if (dtls_transport == NULL) {
GST_WARNING ("Transceiver %p has no DTLS RTCP transport", rtp_trans);
continue;
}
rtcp_transport = dtls_transport->transport;
if (!rtcp_mux && rtcp_transport && rtcp_transport != transport) {
g_object_get (rtcp_transport, "gathering-state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p RTCP gathering state: 0x%x",
rtp_trans, ice_state);
any_state |= (1 << ice_state);
if (ice_state != STATE (COMPLETE))
all_completed = FALSE;
}
}
GST_TRACE_OBJECT (webrtc, "ICE gathering state: 0x%x", any_state);
@ -1064,7 +1025,7 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
g_ptr_array_index (webrtc->priv->transceivers, i);
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
TransportStream *stream = trans->stream;
GstWebRTCDTLSTransport *transport, *rtcp_transport;
GstWebRTCDTLSTransport *transport;
GstWebRTCICEConnectionState ice_state;
GstWebRTCDTLSTransportState dtls_state;
gboolean rtcp_mux = FALSE;
@ -1107,38 +1068,6 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc)
if (ice_state != ICE_STATE (CONNECTED) && ice_state != ICE_STATE (COMPLETED)
&& ice_state != ICE_STATE (CLOSED))
ice_all_connected_completed_or_closed = FALSE;
rtcp_transport = webrtc_transceiver_get_rtcp_dtls_transport (rtp_trans);
if (!rtcp_mux && rtcp_transport && rtcp_transport != transport) {
g_object_get (rtcp_transport, "state", &dtls_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p RTCP DTLS state: 0x%x",
rtp_trans, dtls_state);
any_dtls_state |= (1 << dtls_state);
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CLOSED))
dtls_all_new_or_closed = FALSE;
if (dtls_state != DTLS_STATE (NEW)
&& dtls_state != DTLS_STATE (CONNECTING))
dtls_all_new_connecting_or_checking = FALSE;
if (dtls_state != DTLS_STATE (CONNECTED)
&& dtls_state != DTLS_STATE (CLOSED))
dtls_all_connected_completed_or_closed = FALSE;
g_object_get (rtcp_transport->transport, "state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p RTCP ICE state: 0x%x",
rtp_trans, ice_state);
any_ice_state |= (1 << ice_state);
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CLOSED))
ice_all_new_or_closed = FALSE;
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CHECKING))
ice_all_new_connecting_or_checking = FALSE;
if (ice_state != ICE_STATE (CONNECTED)
&& ice_state != ICE_STATE (COMPLETED)
&& ice_state != ICE_STATE (CLOSED))
ice_all_connected_completed_or_closed = FALSE;
}
}
GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x. DTLS connection "
@ -1704,7 +1633,7 @@ _on_sending_rtcp (GObject * internal_session, GstBuffer * buffer,
pad_name =
g_strdup_printf ("send_rtcp_src_%u",
rtp_trans->sender->rtcp_transport->session_id);
rtp_trans->sender->transport->session_id);
pad = gst_element_get_static_pad (webrtc->rtpbin, pad_name);
g_free (pad_name);
if (pad) {
@ -1895,8 +1824,6 @@ gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc)
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc,
stream->transport->transport);
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc,
stream->rtcp_transport->transport);
}
gst_webrtc_bin_update_sctp_priority (webrtc);
@ -1954,18 +1881,6 @@ _create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
if (webrtc->priv->tos_attached)
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport);
if ((transport = ret->rtcp_transport)) {
g_signal_connect (G_OBJECT (transport->transport),
"notify::state", G_CALLBACK (_on_ice_transport_notify_state), webrtc);
g_signal_connect (G_OBJECT (transport->transport),
"notify::gathering-state",
G_CALLBACK (_on_ice_transport_notify_gathering_state), webrtc);
g_signal_connect (G_OBJECT (transport), "notify::state",
G_CALLBACK (_on_dtls_transport_notify_state), webrtc);
if (webrtc->priv->tos_attached)
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport);
}
GST_TRACE_OBJECT (webrtc,
"Create transport %" GST_PTR_FORMAT " for session %u", ret, session_id);
@ -6985,11 +6900,6 @@ _transport_free (GObject * object)
g_signal_handlers_disconnect_by_data (stream->transport->transport, webrtc);
g_signal_handlers_disconnect_by_data (stream->transport, webrtc);
}
if (stream->rtcp_transport) {
g_signal_handlers_disconnect_by_data (stream->rtcp_transport->transport,
webrtc);
g_signal_handlers_disconnect_by_data (stream->rtcp_transport, webrtc);
}
gst_object_unref (object);
}

4
ext/webrtc/nicetransport.c
View file

@ -171,7 +171,6 @@ gst_webrtc_nice_transport_finalize (GObject * object)
void
gst_webrtc_nice_transport_update_buffer_size (GstWebRTCNiceTransport * nice)
{
GstWebRTCICETransport *ice = GST_WEBRTC_ICE_TRANSPORT (nice);
NiceAgent *agent = NULL;
GPtrArray *sockets;
guint i;
@ -179,8 +178,7 @@ gst_webrtc_nice_transport_update_buffer_size (GstWebRTCNiceTransport * nice)
g_object_get (nice->stream->ice, "agent", &agent, NULL);
g_assert (agent != NULL);
sockets = nice_agent_get_sockets (agent, nice->stream->stream_id,
ice->component + 1);
sockets = nice_agent_get_sockets (agent, nice->stream->stream_id, 1);
if (sockets == NULL) {
g_object_unref (agent);
return;

122
ext/webrtc/transportreceivebin.c
View file

@ -25,24 +25,16 @@
#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 ; ;
* ; '--------' ;
* '---------------------------------------------------------------------------'
* ,-----------------------transport_receive_%u------------------,
* ; ;
* ; ,-nicesrc-, ,-capsfilter-, ,---queue---, ,-dtlssrtpdec-, ;
* ; ; src o-o sink src o-o sink src o-osink rtp_srco---o rtp_src
* ; '---------' '------------' '-----------' ; ; ;
* ; ; rtcp_srco---o rtcp_src
* ; ; ; ;
* ; ; data_srco---o data_src
* ; '-------------' ;
* '-------------------------------------------------------------'
*
* Do we really wnat to be *that* permissive in what we accept?
*
@ -155,24 +147,9 @@ transport_receive_bin_set_receive_state (TransportReceiveBin * receive,
(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);
}
@ -250,9 +227,6 @@ transport_receive_bin_change_state (GstElement * element,
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:
@ -270,9 +244,6 @@ transport_receive_bin_change_state (GstElement * element,
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);
@ -303,7 +274,7 @@ transport_receive_bin_constructed (GObject * object)
TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
GstWebRTCDTLSTransport *transport;
GstPad *ghost, *pad;
GstElement *capsfilter, *funnel, *queue;
GstElement *capsfilter, *queue;
GstCaps *caps;
g_return_if_fail (receive->stream);
@ -336,76 +307,25 @@ transport_receive_bin_constructed (GObject * object)
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");
/* expose rtp_src */
pad =
gst_element_get_static_pad (receive->stream->transport->dtlssrtpdec,
"rtp_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");
/* expose rtcp_rtc */
pad = gst_element_get_static_pad (receive->stream->transport->dtlssrtpdec,
"rtcp_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");
/* expose data_src */
pad = gst_element_get_request_pad (receive->stream->transport->dtlssrtpdec,
"data_src");
ghost = gst_ghost_pad_new ("data_src", pad);
gst_element_add_pad (GST_ELEMENT (receive), ghost);
gst_object_unref (pad);

98
ext/webrtc/transportsendbin.c
View file

@ -25,20 +25,16 @@
#include "utils.h"
/*
* ,------------------------transport_send_%u-------------------------,
* ; ,-----dtlssrtpenc---, ;
* data_sink o--------------------------o data_sink ; ;
* ; ; ; ,---nicesink---, ;
* rtp_sink o--------------------------o rtp_sink_0 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 ; ;
* ; '------------------' '-------------------' '--------------' ;
* '------------------------------------------------------------------'
* ,--------------transport_send_%u-------- ---,
* ; ,-----dtlssrtpenc---, ;
* data_sink o---o data_sink ; ;
* ; ; ; ,---nicesink---, ;
* rtp_sink o---o rtp_sink_0 src o--o sink ; ;
* ; ; ; '--------------' ;
* rtcp_sink o---o rtcp_sink_0 ; ;
* ; '-------------------'
* '-------------------------------------------'
*
* 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?
*/
@ -82,24 +78,6 @@ enum
static void cleanup_blocks (TransportSendBin * send);
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)
@ -112,9 +90,6 @@ transport_send_bin_set_property (GObject * object, guint prop_id,
/* XXX: weak-ref this? Note, it's construct-only so can't be changed later */
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;
@ -133,9 +108,6 @@ transport_send_bin_get_property (GObject * object, guint 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;
@ -201,7 +173,6 @@ transport_send_bin_change_state (GstElement * element,
* we should only add it once/if we get the encoding keys */
TSB_LOCK (send);
gst_element_set_locked_state (send->rtp_ctx.dtlssrtpenc, TRUE);
gst_element_set_locked_state (send->rtcp_ctx.dtlssrtpenc, TRUE);
send->active = TRUE;
TSB_UNLOCK (send);
break;
@ -220,13 +191,6 @@ transport_send_bin_change_state (GstElement * element,
elem = send->stream->transport->transport->sink;
send->rtp_ctx.nice_block = block_peer_pad (elem, "sink");
/* RTCP */
elem = send->stream->rtcp_transport->dtlssrtpenc;
/* Block the RTCP DTLS encoder */
send->rtcp_ctx.rtcp_block = block_peer_pad (elem, "rtcp_sink_0");
/* unblock ice sink once a connection is made, this should also be automatic */
elem = send->stream->rtcp_transport->transport->sink;
send->rtcp_ctx.nice_block = block_peer_pad (elem, "sink");
TSB_UNLOCK (send);
break;
}
@ -257,7 +221,6 @@ transport_send_bin_change_state (GstElement * element,
cleanup_blocks (send);
gst_element_set_locked_state (send->rtp_ctx.dtlssrtpenc, FALSE);
gst_element_set_locked_state (send->rtcp_ctx.dtlssrtpenc, FALSE);
TSB_UNLOCK (send);
break;
@ -276,8 +239,6 @@ _on_dtls_enc_key_set (GstElement * dtlssrtpenc, TransportSendBin * send)
if (dtlssrtpenc == send->rtp_ctx.dtlssrtpenc)
ctx = &send->rtp_ctx;
else if (dtlssrtpenc == send->rtcp_ctx.dtlssrtpenc)
ctx = &send->rtcp_ctx;
else {
GST_WARNING_OBJECT (send,
"Received dtls-enc key info for unknown element %" GST_PTR_FORMAT,
@ -308,8 +269,6 @@ _on_notify_dtls_client_status (GstElement * dtlssrtpenc,
TransportSendBinDTLSContext *ctx;
if (dtlssrtpenc == send->rtp_ctx.dtlssrtpenc)
ctx = &send->rtp_ctx;
else if (dtlssrtpenc == send->rtcp_ctx.dtlssrtpenc)
ctx = &send->rtcp_ctx;
else {
GST_WARNING_OBJECT (send,
"Received dtls-enc client mode for unknown element %" GST_PTR_FORMAT,
@ -351,13 +310,6 @@ _on_notify_ice_connection_state (GstWebRTCICETransport * transport,
_free_pad_block (send->rtp_ctx.nice_block);
send->rtp_ctx.nice_block = NULL;
}
} else if (transport == send->stream->rtcp_transport->transport) {
if (send->rtcp_ctx.nice_block) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtcp_ctx.nice_block->pad);
_free_pad_block (send->rtcp_ctx.nice_block);
send->rtcp_ctx.nice_block = NULL;
}
}
TSB_UNLOCK (send);
}
@ -400,13 +352,6 @@ transport_send_bin_constructed (GObject * object)
g_return_if_fail (send->stream);
g_object_bind_property (send, "rtcp-mux", send->stream, "rtcp-mux",
G_BINDING_BIDIRECTIONAL);
/* Output selector to direct the RTCP for muxed-mode */
send->outputselector = gst_element_factory_make ("output-selector", NULL);
gst_bin_add (GST_BIN (send), send->outputselector);
/* RTP */
transport = send->stream->transport;
/* Do the common init for the context struct */
@ -417,10 +362,6 @@ transport_send_bin_constructed (GObject * object)
pad = gst_element_request_pad (transport->dtlssrtpenc, templ, "rtp_sink_0",
NULL);
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);
@ -436,17 +377,11 @@ transport_send_bin_constructed (GObject * object)
gst_object_unref (pad);
/* RTCP */
transport = send->stream->rtcp_transport;
/* Do the common init for the context struct */
tsb_setup_ctx (send, &send->rtcp_ctx, transport);
templ = _find_pad_template (transport->dtlssrtpenc,
GST_PAD_SINK, GST_PAD_REQUEST, "rtcp_sink_%d");
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");
pad = gst_element_request_pad (transport->dtlssrtpenc, templ, "rtcp_sink_0",
NULL);
ghost = gst_ghost_pad_new ("rtcp_sink", pad);
gst_element_add_pad (GST_ELEMENT (send), ghost);
@ -478,7 +413,6 @@ static void
cleanup_blocks (TransportSendBin * send)
{
cleanup_ctx_blocks (&send->rtp_ctx);
cleanup_ctx_blocks (&send->rtcp_ctx);
}
static void
@ -491,10 +425,6 @@ transport_send_bin_dispose (GObject * object)
g_signal_handlers_disconnect_by_data (send->rtp_ctx.nicesink, send);
send->rtp_ctx.nicesink = NULL;
}
if (send->rtcp_ctx.nicesink) {
g_signal_handlers_disconnect_by_data (send->rtcp_ctx.nicesink, send);
send->rtcp_ctx.nicesink = NULL;
}
cleanup_blocks (send);
TSB_UNLOCK (send);
@ -623,12 +553,6 @@ transport_send_bin_class_init (TransportSendBinClass * klass)
"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

4
ext/webrtc/transportsendbin.h
View file

@ -54,12 +54,8 @@ struct _TransportSendBin
gboolean active; /* Flag that's cleared on shutdown */
TransportStream *stream; /* parent transport stream */
gboolean rtcp_mux;
GstElement *outputselector;
TransportSendBinDTLSContext rtp_ctx;
TransportSendBinDTLSContext rtcp_ctx;
/*
struct pad_block *rtp_block;

19
ext/webrtc/transportstream.c
View file

@ -178,10 +178,6 @@ transport_stream_dispose (GObject * object)
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;
@ -213,19 +209,12 @@ transport_stream_constructed (GObject * 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);
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);
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? */
@ -242,12 +231,6 @@ transport_stream_constructed (GObject * object)
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);

1
ext/webrtc/transportstream.h
View file

@ -58,7 +58,6 @@ struct _TransportStream
GstWebRTCICEStream *stream;
GstWebRTCDTLSTransport *transport;
GstWebRTCDTLSTransport *rtcp_transport;
GArray *ptmap; /* array of PtMapItem's */
GArray *remote_ssrcmap; /* array of SsrcMapItem's */

21
ext/webrtc/webrtctransceiver.c
View file

@ -65,13 +65,6 @@ webrtc_transceiver_set_transport (WebRTCTransceiver * trans,
if (rtp_trans->receiver)
gst_object_replace ((GstObject **) & rtp_trans->receiver->transport,
(GstObject *) stream->transport);
if (rtp_trans->sender)
gst_object_replace ((GstObject **) & rtp_trans->sender->rtcp_transport,
(GstObject *) stream->rtcp_transport);
if (rtp_trans->receiver)
gst_object_replace ((GstObject **) & rtp_trans->receiver->rtcp_transport,
(GstObject *) stream->rtcp_transport);
}
GstWebRTCDTLSTransport *
@ -88,20 +81,6 @@ webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans)
return NULL;
}
GstWebRTCDTLSTransport *
webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans)
{
g_return_val_if_fail (WEBRTC_IS_TRANSCEIVER (trans), NULL);
if (trans->sender) {
return trans->sender->rtcp_transport;
} else if (trans->receiver) {
return trans->receiver->rtcp_transport;
}
return NULL;
}
static void
webrtc_transceiver_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)

23
gst-libs/gst/webrtc/dtlstransport.c
View file

@ -55,8 +55,7 @@ enum
PROP_STATE,
PROP_CLIENT,
PROP_CERTIFICATE,
PROP_REMOTE_CERTIFICATE,
PROP_RTCP,
PROP_REMOTE_CERTIFICATE
};
void
@ -88,9 +87,6 @@ gst_webrtc_dtls_transport_set_property (GObject * object, guint prop_id,
case PROP_CERTIFICATE:
g_object_set_property (G_OBJECT (webrtc->dtlssrtpdec), "pem", value);
break;
case PROP_RTCP:
webrtc->is_rtcp = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -123,9 +119,6 @@ gst_webrtc_dtls_transport_get_property (GObject * object, guint prop_id,
case PROP_REMOTE_CERTIFICATE:
g_object_get_property (G_OBJECT (webrtc->dtlssrtpdec), "peer-pem", value);
break;
case PROP_RTCP:
g_value_set_boolean (value, webrtc->is_rtcp);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -184,8 +177,8 @@ gst_webrtc_dtls_transport_constructed (GObject * object)
/* XXX: this may collide with another connection_id however this is only a
* problem if multiple dtls element sets are being used within the same
* process */
connection_id = g_strdup_printf ("%s_%u_%u", webrtc->is_rtcp ? "rtcp" : "rtp",
webrtc->session_id, g_random_int ());
connection_id = g_strdup_printf ("rtp_%u_%u", webrtc->session_id,
g_random_int ());
webrtc->dtlssrtpenc = gst_element_factory_make ("dtlssrtpenc", NULL);
g_object_set (webrtc->dtlssrtpenc, "connection-id", connection_id,
@ -249,12 +242,6 @@ gst_webrtc_dtls_transport_class_init (GstWebRTCDTLSTransportClass * klass)
g_param_spec_string ("remote-certificate", "Remote DTLS certificate",
"Remote DTLS certificate", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RTCP,
g_param_spec_boolean ("rtcp", "RTCP",
"The transport is being used solely for RTCP", FALSE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
}
static void
@ -263,8 +250,8 @@ gst_webrtc_dtls_transport_init (GstWebRTCDTLSTransport * webrtc)
}
GstWebRTCDTLSTransport *
gst_webrtc_dtls_transport_new (guint session_id, gboolean is_rtcp)
gst_webrtc_dtls_transport_new (guint session_id)
{
return g_object_new (GST_TYPE_WEBRTC_DTLS_TRANSPORT, "session-id", session_id,
"rtcp", is_rtcp, NULL);
NULL);
}

3
gst-libs/gst/webrtc/dtlstransport.h
View file

@ -45,7 +45,6 @@ struct _GstWebRTCDTLSTransport
GstWebRTCICETransport *transport;
GstWebRTCDTLSTransportState state;
gboolean is_rtcp;
gboolean client;
guint session_id;
GstElement *dtlssrtpenc;
@ -62,7 +61,7 @@ struct _GstWebRTCDTLSTransportClass
};
GST_WEBRTC_API
GstWebRTCDTLSTransport * gst_webrtc_dtls_transport_new (guint session_id, gboolean rtcp);
GstWebRTCDTLSTransport * gst_webrtc_dtls_transport_new (guint session_id);
GST_WEBRTC_API
void gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,

11
gst-libs/gst/webrtc/rtpreceiver.c
View file

@ -63,17 +63,6 @@ gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
GST_OBJECT_LOCK (receiver);
gst_object_replace ((GstObject **) & receiver->transport,
GST_OBJECT (transport));
GST_OBJECT_UNLOCK (receiver);
}
void
gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
GST_OBJECT_LOCK (receiver);
gst_object_replace ((GstObject **) & receiver->rtcp_transport,
GST_OBJECT (transport));
GST_OBJECT_UNLOCK (receiver);

3
gst-libs/gst/webrtc/rtpreceiver.h
View file

@ -69,9 +69,6 @@ GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
GST_WEBRTC_API
void gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport);
GST_WEBRTC_API
void gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPReceiver, gst_object_unref)

11
gst-libs/gst/webrtc/rtpsender.c
View file

@ -66,17 +66,6 @@ gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
GST_OBJECT_LOCK (sender);
gst_object_replace ((GstObject **) & sender->transport,
GST_OBJECT (transport));
GST_OBJECT_UNLOCK (sender);
}
void
gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
GST_OBJECT_LOCK (sender);
gst_object_replace ((GstObject **) & sender->rtcp_transport,
GST_OBJECT (transport));
GST_OBJECT_UNLOCK (sender);

3
gst-libs/gst/webrtc/rtpsender.h
View file

@ -83,9 +83,6 @@ GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport);
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport);
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender *sender,
GstWebRTCPriorityType priority);