From df8d29e9c3ced586f9b6a2f3f673b2cd9aca5496 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= Date: Wed, 30 Dec 2020 13:51:21 -0500 Subject: [PATCH] webrtcbin: Remove remnant of non-rtcp-mux mode There was some code left that wasn't used anymore. Part-of: --- ext/webrtc/gstwebrtcbin.c | 17 ++------------- ext/webrtc/gstwebrtcice.c | 39 ---------------------------------- ext/webrtc/transportsendbin.c | 1 - ext/webrtc/transportsendbin.h | 1 - ext/webrtc/transportstream.c | 13 ------------ ext/webrtc/transportstream.h | 3 --- ext/webrtc/webrtctransceiver.h | 1 - 7 files changed, 2 insertions(+), 73 deletions(-) diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index a12e19c700..262b3ee526 100644 --- a/ext/webrtc/gstwebrtcbin.c +++ b/ext/webrtc/gstwebrtcbin.c @@ -949,7 +949,6 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc) GstWebRTCDTLSTransport *dtls_transport; GstWebRTCICETransport *transport; GstWebRTCICEGatheringState ice_state; - gboolean rtcp_mux = FALSE; if (rtp_trans->stopped || stream == NULL) { GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated", @@ -963,8 +962,6 @@ _collate_ice_gathering_states (GstWebRTCBin * webrtc) GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans); } - g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL); - dtls_transport = webrtc_transceiver_get_dtls_transport (rtp_trans); if (dtls_transport == NULL) { GST_WARNING ("Transceiver %p has no DTLS transport", rtp_trans); @@ -1023,12 +1020,9 @@ _collate_peer_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; GstWebRTCDTLSTransport *transport; GstWebRTCICEConnectionState ice_state; GstWebRTCDTLSTransportState dtls_state; - gboolean rtcp_mux = FALSE; if (rtp_trans->stopped) { GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans); @@ -1039,7 +1033,6 @@ _collate_peer_connection_states (GstWebRTCBin * webrtc) continue; } - g_object_get (stream, "rtcp-mux", &rtcp_mux, NULL); transport = webrtc_transceiver_get_dtls_transport (rtp_trans); /* get transport state */ @@ -2137,8 +2130,6 @@ _get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id) webrtc->priv->data_channel_transport = stream; - g_object_set (stream, "rtcp-mux", TRUE, NULL); - if (!(sctp_transport = webrtc->priv->sctp_transport)) { sctp_transport = gst_webrtc_sctp_transport_new (); sctp_transport->transport = @@ -4028,7 +4019,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiverDirection new_dir; const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx); GstWebRTCDTLSSetup new_setup; - gboolean new_rtcp_mux, new_rtcp_rsize; + gboolean new_rtcp_rsize; ReceiveState receive_state = RECEIVE_STATE_UNSET; int i; @@ -4098,8 +4089,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc, } if (!bundled || bundle_idx == media_idx) { - new_rtcp_mux = _media_has_attribute_key (local_media, "rtcp-mux") - && _media_has_attribute_key (remote_media, "rtcp-mux"); new_rtcp_rsize = _media_has_attribute_key (local_media, "rtcp-rsize") && _media_has_attribute_key (remote_media, "rtcp-rsize"); @@ -4112,8 +4101,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc, g_object_unref (session); } } - - g_object_set (stream, "rtcp-mux", new_rtcp_mux, NULL); } } @@ -4319,7 +4306,7 @@ _update_data_channel_from_sdp_media (GstWebRTCBin * webrtc, /* data channel is always rtcp-muxed to avoid generating ICE candidates * for RTCP */ - g_object_set (stream, "rtcp-mux", TRUE, "dtls-client", + g_object_set (stream, "dtls-client", new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL); local_port = _get_sctp_port_from_media (local_media); diff --git a/ext/webrtc/gstwebrtcice.c b/ext/webrtc/gstwebrtcice.c index 4a2ae33c57..c780ab2bb5 100644 --- a/ext/webrtc/gstwebrtcice.c +++ b/ext/webrtc/gstwebrtcice.c @@ -138,35 +138,6 @@ _stop_thread (GstWebRTCICE * ice) g_thread_unref (ice->priv->thread); } -#if 0 -static NiceComponentType -_webrtc_component_to_nice (GstWebRTCICEComponent comp) -{ - switch (comp) { - case GST_WEBRTC_ICE_COMPONENT_RTP: - return NICE_COMPONENT_TYPE_RTP; - case GST_WEBRTC_ICE_COMPONENT_RTCP: - return NICE_COMPONENT_TYPE_RTCP; - default: - g_assert_not_reached (); - return 0; - } -} - -static GstWebRTCICEComponent -_nice_component_to_webrtc (NiceComponentType comp) -{ - switch (comp) { - case NICE_COMPONENT_TYPE_RTP: - return GST_WEBRTC_ICE_COMPONENT_RTP; - case NICE_COMPONENT_TYPE_RTCP: - return GST_WEBRTC_ICE_COMPONENT_RTCP; - default: - g_assert_not_reached (); - return 0; - } -} -#endif struct NiceStreamItem { guint session_id; @@ -380,16 +351,6 @@ _add_turn_server (GstWebRTCICE * ice, struct NiceStreamItem *item, g_free (uri); break; } - ret = nice_agent_set_relay_info (ice->priv->nice_agent, - item->nice_stream_id, NICE_COMPONENT_TYPE_RTCP, - gst_uri_get_host (turn_server), gst_uri_get_port (turn_server), - user, pass, relays[i]); - if (!ret) { - gchar *uri = gst_uri_to_string (turn_server); - GST_ERROR_OBJECT (ice, "Failed to set TURN server '%s'", uri); - g_free (uri); - break; - } } g_free (user); g_free (pass); diff --git a/ext/webrtc/transportsendbin.c b/ext/webrtc/transportsendbin.c index 7484a2ff7d..5bc611d3c2 100644 --- a/ext/webrtc/transportsendbin.c +++ b/ext/webrtc/transportsendbin.c @@ -69,7 +69,6 @@ enum { PROP_0, PROP_STREAM, - PROP_RTCP_MUX, }; #define TSB_GET_LOCK(tsb) (&tsb->lock) diff --git a/ext/webrtc/transportsendbin.h b/ext/webrtc/transportsendbin.h index 8bc688ddaa..498997231c 100644 --- a/ext/webrtc/transportsendbin.h +++ b/ext/webrtc/transportsendbin.h @@ -59,7 +59,6 @@ struct _TransportSendBin /* struct pad_block *rtp_block; - struct pad_block *rtcp_mux_block; struct pad_block *rtp_nice_block; struct pad_block *rtcp_block; diff --git a/ext/webrtc/transportstream.c b/ext/webrtc/transportstream.c index 11e9cad8a8..c08bfc0568 100644 --- a/ext/webrtc/transportstream.c +++ b/ext/webrtc/transportstream.c @@ -36,7 +36,6 @@ enum PROP_0, PROP_WEBRTC, PROP_SESSION_ID, - PROP_RTCP_MUX, PROP_DTLS_CLIENT, }; @@ -124,9 +123,6 @@ transport_stream_set_property (GObject * object, guint prop_id, 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; @@ -148,9 +144,6 @@ transport_stream_get_property (GObject * object, guint 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; @@ -270,12 +263,6 @@ transport_stream_class_init (TransportStreamClass * klass) 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", diff --git a/ext/webrtc/transportstream.h b/ext/webrtc/transportstream.h index da05c2207d..4642c9a73c 100644 --- a/ext/webrtc/transportstream.h +++ b/ext/webrtc/transportstream.h @@ -52,9 +52,6 @@ struct _TransportStream GstObject parent; guint session_id; /* session_id */ - gboolean rtcp; - gboolean rtcp_mux; - gboolean rtcp_rsize; gboolean dtls_client; gboolean active; /* TRUE if any mline in the bundle/transport is active */ TransportSendBin *send_bin; /* bin containing all the sending transport elements */ diff --git a/ext/webrtc/webrtctransceiver.h b/ext/webrtc/webrtctransceiver.h index 6d2d963eff..2f65b75593 100644 --- a/ext/webrtc/webrtctransceiver.h +++ b/ext/webrtc/webrtctransceiver.h @@ -63,7 +63,6 @@ void webrtc_transceiver_set_transport (WebRTCTransceiver * TransportStream * stream); GstWebRTCDTLSTransport * webrtc_transceiver_get_dtls_transport (GstWebRTCRTPTransceiver * trans); -GstWebRTCDTLSTransport * webrtc_transceiver_get_rtcp_dtls_transport (GstWebRTCRTPTransceiver * trans); G_END_DECLS