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webrtc: add new post-rtp-aux-sender signal

Browse source 
Adds a new signal to webrtcbin, to allow for placement
of an object after rtp, before sendbin. This is usable for
objects such as congestion control elements, that don't want
to be burdened by the synchronization requirements of rtpsession.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7940>
This commit is contained in:
Albert Sjolund 2024-11-22 12:46:22 +01:00 committed by GStreamer Marge Bot
parent 90d99c7b52
commit 97a3bba629
2 changed files with 80 additions and 10 deletions
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10
subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json
View file

@ -253815,6 +253815,16 @@
"return-type": "GstElement",
"when": "last"
},
"request-post-rtp-aux-sender": {
"args": [
{
"name": "arg0",
"type": "GstWebRTCDTLSTransport"
}
],
"return-type": "GstElement",
"when": "last"
},
"set-local-description": {
"action": true,
"args": [

80
subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
View file

@ -654,6 +654,7 @@ enum
ON_DATA_CHANNEL_SIGNAL,
PREPARE_DATA_CHANNEL_SIGNAL,
REQUEST_AUX_SENDER,
REQUEST_POST_RTP_AUX_SENDER,
ADD_ICE_CANDIDATE_FULL_SIGNAL,
LAST_SIGNAL,
};
@ -5064,6 +5065,56 @@ try_match_transceiver_with_fec_decoder (GstWebRTCBin * webrtc,
}
}
/* connects a specific sessions send_rtp_src pad with the matching sink pad in
* the transport stream. If an aux sender was requested, insert it between these
* elements, before linking. */
static void
connect_rtpbin_with_sendbin (GstWebRTCBin * webrtc, guint session_id,
TransportStream * stream)
{
GstElement *aux_sender;
g_signal_emit (webrtc,
gst_webrtc_bin_signals[REQUEST_POST_RTP_AUX_SENDER], 0,
stream->transport, &aux_sender);
gboolean aux_success = FALSE;
gchar *rtp_pad_name = g_strdup_printf ("send_rtp_src_%u", session_id);
if (aux_sender) {
gst_object_ref_sink (aux_sender);
if (!gst_bin_add (GST_BIN (webrtc), aux_sender)) {
GST_ERROR_OBJECT (webrtc,
"Unable to add aux_sender %" GST_PTR_FORMAT " to webrtcbin"
". Skipping it.", aux_sender);
goto aux_done;
}
gst_element_sync_state_with_parent (aux_sender);
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), rtp_pad_name,
aux_sender, "sink")) {
GST_ERROR_OBJECT (webrtc,
"Unable to link aux_sender %" GST_PTR_FORMAT " to %" GST_PTR_FORMAT
". Skipping it.", webrtc->rtpbin, aux_sender);
goto aux_done;
}
if (!gst_element_link_pads (aux_sender, "src",
GST_ELEMENT (stream->send_bin), "rtp_sink")) {
gst_element_unlink_pads (GST_ELEMENT (webrtc->rtpbin), rtp_pad_name,
aux_sender, "sink");
GST_ERROR_OBJECT (webrtc,
"Unable to link %" GST_PTR_FORMAT " to aux sender %" GST_PTR_FORMAT
". Skipping it.", aux_sender, stream->send_bin);
goto aux_done;
}
aux_success = TRUE;
aux_done:
gst_clear_object (&aux_sender);
}
if (!aux_success) {
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), rtp_pad_name,
GST_ELEMENT (stream->send_bin), "rtp_sink"))
g_warn_if_reached ();
}
g_free (rtp_pad_name);
}
static void
_set_internal_rtpbin_element_props_from_stream (GstWebRTCBin * webrtc,
TransportStream * stream)
@ -5269,11 +5320,7 @@ _connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
gst_pad_link (srcpad, rtp_sink);
gst_object_unref (rtp_sink);
pad_name = g_strdup_printf ("send_rtp_src_%u", pad->trans->mline);
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
GST_ELEMENT (trans->stream->send_bin), "rtp_sink"))
g_warn_if_reached ();
g_free (pad_name);
connect_rtpbin_with_sendbin (webrtc, pad->trans->mline, trans->stream);
} else {
gchar *pad_name = g_strdup_printf ("sink_%u", pad->trans->mline);
GstPad *funnel_sinkpad =
@ -6082,11 +6129,7 @@ _connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id, GError ** error)
gst_object_unref (srcpad);
gst_object_unref (rtp_sink);
pad_name = g_strdup_printf ("send_rtp_src_%d", session_id);
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
GST_ELEMENT (stream->send_bin), "rtp_sink"))
g_warn_if_reached ();
g_free (pad_name);
connect_rtpbin_with_sendbin (webrtc, session_id, stream);
done:
return TRUE;
@ -9280,6 +9323,23 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
G_SIGNAL_RUN_LAST, 0, _gst_element_accumulator, NULL, NULL,
GST_TYPE_ELEMENT, 1, GST_TYPE_WEBRTC_DTLS_TRANSPORT);
/**
* GstWebRTCBin::request-post-rtp-aux-sender:
* @object: the #GstWebRTCBin
* @dtls-transport: The #GstWebRTCDTLSTransport object for which the aux
* sender will be used.
*
* Request an AUX sender element for the given @dtls-transport,
* that is placed between RTPBin and Transportbin..
*
* Returns: (transfer full): A new GStreamer element
*
* Since: 1.26
*/
gst_webrtc_bin_signals[REQUEST_POST_RTP_AUX_SENDER] =
g_signal_new ("request-post-rtp-aux-sender",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, 0, _gst_element_accumulator,
NULL, NULL, GST_TYPE_ELEMENT, 1, GST_TYPE_WEBRTC_DTLS_TRANSPORT);
/**
* GstWebRTCBin::add-transceiver:
* @object: the #webrtcbin