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webrtcbin: Attach rtpbin even for data channels

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This is required because the same transport may later be used for RTP.
In which case the RTCP needs to flow bi-directionnally already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2172>
This commit is contained in:
Olivier Crête 2021-04-16 20:39:35 -04:00 committed by GStreamer Marge Bot
parent 6f2c010360
commit c690be3e03
1 changed files with 19 additions and 25 deletions
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44
ext/webrtc/gstwebrtcbin.c
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@ -1893,6 +1893,7 @@ _create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
{ {
GstWebRTCDTLSTransport *transport; GstWebRTCDTLSTransport *transport;
TransportStream *ret; TransportStream *ret;
gchar *pad_name;
/* FIXME: how to parametrize the sender and the receiver */ /* FIXME: how to parametrize the sender and the receiver */
ret = transport_stream_new (webrtc, session_id); ret = transport_stream_new (webrtc, session_id);
@ -1908,6 +1909,22 @@ _create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
if (webrtc->priv->tos_attached) if (webrtc->priv->tos_attached)
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport); gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport);
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->send_bin));
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->receive_bin));
g_ptr_array_add (webrtc->priv->transports, ret);
pad_name = g_strdup_printf ("recv_rtcp_sink_%u", ret->session_id);
if (!gst_element_link_pads (GST_ELEMENT (ret->receive_bin), "rtcp_src",
GST_ELEMENT (webrtc->rtpbin), pad_name))
g_warn_if_reached ();
g_free (pad_name);
pad_name = g_strdup_printf ("send_rtcp_src_%u", ret->session_id);
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
GST_ELEMENT (ret->send_bin), "rtcp_sink"))
g_warn_if_reached ();
g_free (pad_name);
GST_TRACE_OBJECT (webrtc, GST_TRACE_OBJECT (webrtc,
"Create transport %" GST_PTR_FORMAT " for session %u", ret, session_id); "Create transport %" GST_PTR_FORMAT " for session %u", ret, session_id);
@ -1918,30 +1935,11 @@ static TransportStream *
_get_or_create_rtp_transport_channel (GstWebRTCBin * webrtc, guint session_id) _get_or_create_rtp_transport_channel (GstWebRTCBin * webrtc, guint session_id)
{ {
TransportStream *ret; TransportStream *ret;
gchar *pad_name;
ret = _find_transport_for_session (webrtc, session_id); ret = _find_transport_for_session (webrtc, session_id);
if (!ret) { if (!ret)
ret = _create_transport_channel (webrtc, session_id); ret = _create_transport_channel (webrtc, session_id);
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->send_bin));
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->receive_bin));
g_ptr_array_add (webrtc->priv->transports, ret);
pad_name = g_strdup_printf ("recv_rtcp_sink_%u", ret->session_id);
if (!gst_element_link_pads (GST_ELEMENT (ret->receive_bin), "rtcp_src",
GST_ELEMENT (webrtc->rtpbin), pad_name))
g_warn_if_reached ();
g_free (pad_name);
pad_name = g_strdup_printf ("send_rtcp_src_%u", ret->session_id);
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
GST_ELEMENT (ret->send_bin), "rtcp_sink"))
g_warn_if_reached ();
if (webrtc->priv->tos_attached)
gst_webrtc_bin_attach_tos_to_session (webrtc, ret->session_id);
g_free (pad_name);
}
gst_element_sync_state_with_parent (GST_ELEMENT (ret->send_bin)); gst_element_sync_state_with_parent (GST_ELEMENT (ret->send_bin));
gst_element_sync_state_with_parent (GST_ELEMENT (ret->receive_bin)); gst_element_sync_state_with_parent (GST_ELEMENT (ret->receive_bin));
@ -2157,12 +2155,8 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
stream = _find_transport_for_session (webrtc, session_id); stream = _find_transport_for_session (webrtc, session_id);
if (!stream) { if (!stream)
stream = _create_transport_channel (webrtc, session_id); stream = _create_transport_channel (webrtc, session_id);
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (stream->send_bin));
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (stream->receive_bin));
g_ptr_array_add (webrtc->priv->transports, stream);
}
webrtc->priv->data_channel_transport = stream; webrtc->priv->data_channel_transport = stream;