From 073df3d82094be9c8ed397f9e96146d9a205d0f9 Mon Sep 17 00:00:00 2001 From: Thibault Saunier Date: Sat, 4 Jun 2022 18:07:16 -0400 Subject: [PATCH] webrtcbin: Add a signal to plug bandwidth estimator elements We need GStreamer elements to do the bandwidth estimation as this way they can also control the pacing of the transmission flow as specified in the [GCC] algorithm for example. Bandwidth estimator element are placed right before the "RTPSession" as an "rtp-aux-sender" element. This way they can use the "Transport-wide Congestion Control" RTCP feedback messages through the "RTPTwcc" custom events that are sent by the rtpsession. Applications are responsible to react to the bandwidth estimator element and set the encoder target bitrate etc... which means that we can not pass an estimator as an element factory, so a signal as been chosen instead. [GCC]: https://datatracker.ietf.org/doc/html/draft-ietf-rmcat-gcc-02 Part-of: --- .../docs/plugins/gst_plugins_cache.json | 10 +++ .../gst-plugins-bad/ext/webrtc/gstwebrtcbin.c | 81 +++++++++++++++++++ 2 files changed, 91 insertions(+) diff --git a/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json b/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json index b0e76682e4..2b992cafb2 100644 --- a/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json +++ b/subprojects/gst-plugins-bad/docs/plugins/gst_plugins_cache.json @@ -231228,6 +231228,16 @@ "return-type": "void", "when": "last" }, + "request-aux-sender": { + "args": [ + { + "name": "arg0", + "type": "GstWebRTCDTLSTransport" + } + ], + "return-type": "GstElement", + "when": "last" + }, "set-local-description": { "action": true, "args": [ diff --git a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c index 4d4e6b892c..6481724584 100644 --- a/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c +++ b/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c @@ -216,6 +216,21 @@ _have_dtls_elements (GstWebRTCBin * webrtc) return TRUE; } +static gboolean +_gst_element_accumulator (GSignalInvocationHint * ihint, + GValue * return_accu, const GValue * handler_return, gpointer dummy) +{ + GstElement *element; + + element = g_value_get_object (handler_return); + GST_DEBUG ("got element %" GST_PTR_FORMAT, element); + + g_value_set_object (return_accu, element); + + /* stop emission if we have an element */ + return (element == NULL); +} + G_DEFINE_TYPE (GstWebRTCBinPad, gst_webrtc_bin_pad, GST_TYPE_GHOST_PAD); static void @@ -494,6 +509,7 @@ enum CREATE_DATA_CHANNEL_SIGNAL, ON_DATA_CHANNEL_SIGNAL, PREPARE_DATA_CHANNEL_SIGNAL, + REQUEST_AUX_SENDER, LAST_SIGNAL, }; @@ -7107,6 +7123,7 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id, GstElement *ret, *rtx; GstPad *pad; char *name; + GstElement *aux_sender = NULL; stream = _find_transport_for_session (webrtc, session_id); if (!stream) { @@ -7138,6 +7155,53 @@ on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id, name = g_strdup_printf ("src_%u", session_id); pad = gst_element_get_static_pad (rtx, "src"); + + + g_signal_emit (webrtc, gst_webrtc_bin_signals[REQUEST_AUX_SENDER], 0, + stream->transport, &aux_sender); + if (aux_sender) { + GstPadLinkReturn link_res; + GstPad *sinkpad = gst_element_get_static_pad (aux_sender, "sink"); + GstPad *srcpad = gst_element_get_static_pad (aux_sender, "src"); + + gst_object_ref_sink (aux_sender); + + if (!sinkpad || !srcpad) { + GST_ERROR_OBJECT (webrtc, + "Invalid pads for the aux sender %" GST_PTR_FORMAT + ". Skipping it.", aux_sender); + goto bwe_done; + } + + if (!gst_bin_add (GST_BIN (ret), aux_sender)) { + GST_ERROR_OBJECT (webrtc, + "Could not add aux sender %" GST_PTR_FORMAT, aux_sender); + goto bwe_done; + } + + link_res = gst_pad_link (pad, sinkpad); + if (link_res != GST_PAD_LINK_OK) { + GST_ERROR_OBJECT (webrtc, + "Could not link aux sender %" GST_PTR_FORMAT " %s", aux_sender, + gst_pad_link_get_name (link_res)); + goto bwe_done; + } + + gst_clear_object (&pad); + pad = gst_object_ref (srcpad); + + bwe_done: + if (pad != srcpad) { + /* Failed using the provided aux sender */ + if (gst_object_has_as_parent (GST_OBJECT (aux_sender), GST_OBJECT (ret))) { + gst_bin_remove (GST_BIN (ret), aux_sender); + } + } + gst_clear_object (&aux_sender); + gst_clear_object (&srcpad); + gst_clear_object (&sinkpad); + } + if (!gst_element_add_pad (ret, gst_ghost_pad_new (name, pad))) g_warn_if_reached (); gst_clear_object (&pad); @@ -8574,6 +8638,23 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass) G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 2, GST_TYPE_WEBRTC_DATA_CHANNEL, G_TYPE_BOOLEAN); + /** + * GstWebRTCBin::request-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. + * + * Returns: (transfer full): A new GStreamer element + * + * Since: 1.22 + */ + gst_webrtc_bin_signals[REQUEST_AUX_SENDER] = + g_signal_new ("request-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