mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-03 16:09:39 +00:00
3bdf1e691e
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2260>
536 lines
16 KiB
C
536 lines
16 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "transportsendbin.h"
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#include "utils.h"
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/*
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* ,--------------transport_send_%u-------- ---,
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* ; ,-----dtlssrtpenc---, ;
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* data_sink o---o data_sink ; ;
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* ; ; ; ,---nicesink---, ;
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* rtp_sink o---o rtp_sink_0 src o--o sink ; ;
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* ; ; ; '--------------' ;
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* rtcp_sink o---o rtcp_sink_0 ; ;
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* ; '-------------------'
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* '-------------------------------------------'
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*
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*
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* FIXME: Do we need a valve drop=TRUE for the no RTCP case?
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*/
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#define GST_CAT_DEFAULT gst_webrtc_transport_send_bin_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define transport_send_bin_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (TransportSendBin, transport_send_bin, GST_TYPE_BIN,
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_transport_send_bin_debug,
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"webrtctransportsendbin", 0, "webrtctransportsendbin"););
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static GstStaticPadTemplate rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtp_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstStaticPadTemplate rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstStaticPadTemplate data_sink_template =
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GST_STATIC_PAD_TEMPLATE ("data_sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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enum
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{
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PROP_0,
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PROP_STREAM,
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};
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#define TSB_GET_LOCK(tsb) (&tsb->lock)
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#define TSB_LOCK(tsb) (g_mutex_lock (TSB_GET_LOCK(tsb)))
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#define TSB_UNLOCK(tsb) (g_mutex_unlock (TSB_GET_LOCK(tsb)))
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static void cleanup_blocks (TransportSendBin * send);
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static void
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transport_send_bin_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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TransportSendBin *send = TRANSPORT_SEND_BIN (object);
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GST_OBJECT_LOCK (send);
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switch (prop_id) {
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case PROP_STREAM:
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/* XXX: weak-ref this? Note, it's construct-only so can't be changed later */
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send->stream = TRANSPORT_STREAM (g_value_get_object (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (send);
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}
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static void
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transport_send_bin_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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TransportSendBin *send = TRANSPORT_SEND_BIN (object);
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GST_OBJECT_LOCK (send);
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switch (prop_id) {
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case PROP_STREAM:
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g_value_set_object (value, send->stream);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (send);
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}
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static GstPadProbeReturn
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pad_block (GstPad * pad, GstPadProbeInfo * info, gpointer unused)
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{
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/* Drop all events: we don't care about them and don't want to block on
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* them. Sticky events would be forwarded again later once we unblock
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* and we don't want to forward them here already because that might
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* cause a spurious GST_FLOW_FLUSHING */
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if (GST_IS_EVENT (info->data))
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return GST_PAD_PROBE_DROP;
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/* But block on any actual data-flow so we don't accidentally send that
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* to a pad that is not ready yet, causing GST_FLOW_FLUSHING and everything
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* to silently stop.
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*/
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GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data);
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return GST_PAD_PROBE_OK;
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}
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/* We block RTP/RTCP dataflow until the relevant DTLS key
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* nego is done, but we need to block the *peer* src pad
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* because the dtlssrtpenc state changes are done manually,
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* and otherwise we can get state change problems trying to shut down */
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static struct pad_block *
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block_peer_pad (GstElement * elem, const gchar * pad_name)
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{
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GstPad *pad, *peer;
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struct pad_block *block;
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pad = gst_element_get_static_pad (elem, pad_name);
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peer = gst_pad_get_peer (pad);
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block = _create_pad_block (elem, peer, 0, NULL, NULL);
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block->block_id = gst_pad_add_probe (peer,
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GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM,
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(GstPadProbeCallback) pad_block, NULL, NULL);
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gst_object_unref (pad);
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gst_object_unref (peer);
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return block;
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}
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static GstStateChangeReturn
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transport_send_bin_change_state (GstElement * element,
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GstStateChange transition)
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{
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TransportSendBin *send = TRANSPORT_SEND_BIN (element);
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GST_DEBUG_OBJECT (element, "changing state: %s => %s",
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gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
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gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:{
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/* XXX: don't change state until the client-ness has been chosen
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* arguably the element should be able to deal with this itself or
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* we should only add it once/if we get the encoding keys */
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TSB_LOCK (send);
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gst_element_set_locked_state (send->dtlssrtpenc, TRUE);
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send->active = TRUE;
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send->has_clientness = FALSE;
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TSB_UNLOCK (send);
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break;
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}
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case GST_STATE_CHANGE_READY_TO_PAUSED:{
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GstElement *elem;
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TSB_LOCK (send);
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/* RTP */
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/* unblock the encoder once the key is set, this should also be automatic */
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elem = send->stream->transport->dtlssrtpenc;
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send->rtp_block = block_peer_pad (elem, "rtp_sink_0");
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/* Also block the RTCP pad on the RTP encoder, in case we mux RTCP */
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send->rtcp_block = block_peer_pad (elem, "rtcp_sink_0");
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/* unblock ice sink once a connection is made, this should also be automatic */
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elem = send->stream->transport->transport->sink;
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TSB_UNLOCK (send);
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break;
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}
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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if (ret == GST_STATE_CHANGE_FAILURE) {
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GST_WARNING_OBJECT (element, "Parent state change handler failed");
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return ret;
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}
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switch (transition) {
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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{
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/* Now that everything is stopped, we can remove the pad blocks
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* if they still exist, without accidentally feeding data to the
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* dtlssrtpenc elements */
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TSB_LOCK (send);
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cleanup_blocks (send);
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TSB_UNLOCK (send);
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break;
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}
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case GST_STATE_CHANGE_READY_TO_NULL:{
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TSB_LOCK (send);
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send->active = FALSE;
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cleanup_blocks (send);
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gst_element_set_locked_state (send->dtlssrtpenc, FALSE);
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TSB_UNLOCK (send);
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break;
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}
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default:
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break;
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}
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return ret;
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}
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static void
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_on_dtls_enc_key_set (GstElement * dtlssrtpenc, TransportSendBin * send)
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{
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if (dtlssrtpenc != send->dtlssrtpenc) {
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GST_WARNING_OBJECT (send,
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"Received dtls-enc key info for unknown element %" GST_PTR_FORMAT,
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dtlssrtpenc);
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return;
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}
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TSB_LOCK (send);
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if (!send->active) {
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GST_INFO_OBJECT (send, "Received dtls-enc key info from %" GST_PTR_FORMAT
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"when not active", dtlssrtpenc);
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goto done;
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}
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GST_LOG_OBJECT (send, "Unblocking %" GST_PTR_FORMAT " pads", dtlssrtpenc);
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_free_pad_block (send->rtp_block);
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_free_pad_block (send->rtcp_block);
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send->rtp_block = send->rtcp_block = NULL;
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done:
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TSB_UNLOCK (send);
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}
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static void
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maybe_start_enc (TransportSendBin * send)
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{
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GstWebRTCICEConnectionState state;
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if (!send->has_clientness) {
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GST_LOG_OBJECT (send, "Can't start DTLS because doesn't know client-ness");
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return;
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}
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g_object_get (send->stream->transport->transport, "state", &state, NULL);
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if (state != GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED &&
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state != GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED) {
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GST_LOG_OBJECT (send, "Can't start DTLS yet because ICE is not connected.");
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return;
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}
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gst_element_set_locked_state (send->dtlssrtpenc, FALSE);
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gst_element_sync_state_with_parent (send->dtlssrtpenc);
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}
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static void
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_on_notify_dtls_client_status (GstElement * dtlssrtpenc,
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GParamSpec * pspec, TransportSendBin * send)
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{
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if (dtlssrtpenc != send->dtlssrtpenc) {
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GST_WARNING_OBJECT (send,
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"Received dtls-enc client mode for unknown element %" GST_PTR_FORMAT,
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dtlssrtpenc);
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return;
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}
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TSB_LOCK (send);
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if (!send->active) {
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GST_DEBUG_OBJECT (send,
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"DTLS-SRTP encoder ready after we're already stopping");
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goto done;
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}
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send->has_clientness = TRUE;
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GST_DEBUG_OBJECT (send,
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"DTLS-SRTP encoder configured. Unlocking it and maybe changing state %"
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GST_PTR_FORMAT, dtlssrtpenc);
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maybe_start_enc (send);
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done:
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TSB_UNLOCK (send);
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}
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static void
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_on_notify_ice_connection_state (GstWebRTCICETransport * transport,
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GParamSpec * pspec, TransportSendBin * send)
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{
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TSB_LOCK (send);
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maybe_start_enc (send);
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TSB_UNLOCK (send);
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}
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static void
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transport_send_bin_constructed (GObject * object)
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{
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TransportSendBin *send = TRANSPORT_SEND_BIN (object);
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GstPadTemplate *templ;
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GstPad *ghost, *pad;
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g_return_if_fail (send->stream);
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send->dtlssrtpenc = send->stream->transport->dtlssrtpenc;
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send->nicesink = send->stream->transport->transport->sink;
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/* unblock the encoder once the key is set */
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g_signal_connect (send->dtlssrtpenc, "on-key-set",
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G_CALLBACK (_on_dtls_enc_key_set), send);
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/* Bring the encoder up to current state only once the is-client prop is set */
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g_signal_connect (send->dtlssrtpenc, "notify::is-client",
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G_CALLBACK (_on_notify_dtls_client_status), send);
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/* unblock ice sink once it signals a connection */
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g_signal_connect (send->stream->transport->transport, "notify::state",
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G_CALLBACK (_on_notify_ice_connection_state), send);
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gst_bin_add (GST_BIN (send), GST_ELEMENT (send->dtlssrtpenc));
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gst_bin_add (GST_BIN (send), GST_ELEMENT (send->nicesink));
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if (!gst_element_link_pads (GST_ELEMENT (send->dtlssrtpenc), "src",
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send->nicesink, "sink"))
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g_warn_if_reached ();
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templ = _find_pad_template (send->dtlssrtpenc, GST_PAD_SINK, GST_PAD_REQUEST,
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"rtp_sink_%d");
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pad = gst_element_request_pad (send->dtlssrtpenc, templ, "rtp_sink_0", NULL);
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ghost = gst_ghost_pad_new ("rtp_sink", pad);
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gst_element_add_pad (GST_ELEMENT (send), ghost);
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gst_object_unref (pad);
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/* push the data stream onto the RTP dtls element */
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templ = _find_pad_template (send->dtlssrtpenc, GST_PAD_SINK, GST_PAD_REQUEST,
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"data_sink");
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pad = gst_element_request_pad (send->dtlssrtpenc, templ, "data_sink", NULL);
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ghost = gst_ghost_pad_new ("data_sink", pad);
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gst_element_add_pad (GST_ELEMENT (send), ghost);
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gst_object_unref (pad);
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/* RTCP */
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/* Do the common init for the context struct */
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templ = _find_pad_template (send->dtlssrtpenc, GST_PAD_SINK, GST_PAD_REQUEST,
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"rtcp_sink_%d");
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pad = gst_element_request_pad (send->dtlssrtpenc, templ, "rtcp_sink_0", NULL);
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ghost = gst_ghost_pad_new ("rtcp_sink", pad);
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gst_element_add_pad (GST_ELEMENT (send), ghost);
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gst_object_unref (pad);
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G_OBJECT_CLASS (parent_class)->constructed (object);
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}
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static void
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cleanup_blocks (TransportSendBin * send)
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{
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if (send->rtp_block) {
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_free_pad_block (send->rtp_block);
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send->rtp_block = NULL;
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}
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if (send->rtcp_block) {
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_free_pad_block (send->rtcp_block);
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send->rtcp_block = NULL;
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}
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}
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static void
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transport_send_bin_dispose (GObject * object)
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{
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TransportSendBin *send = TRANSPORT_SEND_BIN (object);
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TSB_LOCK (send);
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if (send->nicesink) {
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g_signal_handlers_disconnect_by_data (send->nicesink, send);
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send->nicesink = NULL;
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}
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cleanup_blocks (send);
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TSB_UNLOCK (send);
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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transport_send_bin_finalize (GObject * object)
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{
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TransportSendBin *send = TRANSPORT_SEND_BIN (object);
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g_mutex_clear (TSB_GET_LOCK (send));
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_transport_send_bin_element_query (GstElement * element, GstQuery * query)
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{
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gboolean ret = TRUE;
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GstClockTime min_latency;
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GST_LOG_OBJECT (element, "got query %s", GST_QUERY_TYPE_NAME (query));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_LATENCY:
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/* when latency is queried, use the result to configure our
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* own latency internally, piggybacking off the global
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* latency configuration sequence. */
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GST_DEBUG_OBJECT (element, "handling latency query");
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/* Call the parent query handler to actually get the query
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* sent upstream */
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ret =
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GST_ELEMENT_CLASS (transport_send_bin_parent_class)->query
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(GST_ELEMENT (element), query);
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if (!ret)
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break;
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gst_query_parse_latency (query, NULL, &min_latency, NULL);
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GST_DEBUG_OBJECT (element,
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"got min latency %" GST_TIME_FORMAT, GST_TIME_ARGS (min_latency));
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/* configure latency on elements */
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/* Call the parent event handler, because our sub-class handler
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* will drop the LATENCY event. We also don't need to that
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* the latency configuration is valid (min < max), because
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* the pipeline will do it when checking the query results */
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if (GST_ELEMENT_CLASS (transport_send_bin_parent_class)->send_event
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(GST_ELEMENT (element), gst_event_new_latency (min_latency))) {
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GST_INFO_OBJECT (element, "configured latency of %" GST_TIME_FORMAT,
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GST_TIME_ARGS (min_latency));
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} else {
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GST_WARNING_OBJECT (element,
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"did not really configure latency of %" GST_TIME_FORMAT,
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GST_TIME_ARGS (min_latency));
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}
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break;
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default:
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ret =
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GST_ELEMENT_CLASS (transport_send_bin_parent_class)->query
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(GST_ELEMENT (element), query);
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break;
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}
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return ret;
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}
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static gboolean
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gst_transport_send_bin_element_event (GstElement * element, GstEvent * event)
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{
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gboolean ret = TRUE;
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GST_LOG_OBJECT (element, "got event %s", GST_EVENT_TYPE_NAME (event));
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_LATENCY:
|
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/* Ignore the pipeline configured latency, we choose our own
|
|
* instead when the latency query happens, so that sending
|
|
* isn't affected by other parts of the pipeline */
|
|
GST_DEBUG_OBJECT (element, "Ignoring latency event from parent");
|
|
gst_event_unref (event);
|
|
break;
|
|
default:
|
|
ret =
|
|
GST_ELEMENT_CLASS (transport_send_bin_parent_class)->send_event
|
|
(GST_ELEMENT (element), event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
transport_send_bin_class_init (TransportSendBinClass * klass)
|
|
{
|
|
GObjectClass *gobject_class = (GObjectClass *) klass;
|
|
GstElementClass *element_class = (GstElementClass *) klass;
|
|
|
|
element_class->change_state = transport_send_bin_change_state;
|
|
|
|
gst_element_class_add_static_pad_template (element_class, &rtp_sink_template);
|
|
gst_element_class_add_static_pad_template (element_class,
|
|
&rtcp_sink_template);
|
|
gst_element_class_add_static_pad_template (element_class,
|
|
&data_sink_template);
|
|
|
|
gst_element_class_set_metadata (element_class, "WebRTC Transport Send Bin",
|
|
"Filter/Network/WebRTC", "A bin for webrtc connections",
|
|
"Matthew Waters <matthew@centricular.com>");
|
|
|
|
gobject_class->constructed = transport_send_bin_constructed;
|
|
gobject_class->dispose = transport_send_bin_dispose;
|
|
gobject_class->get_property = transport_send_bin_get_property;
|
|
gobject_class->set_property = transport_send_bin_set_property;
|
|
gobject_class->finalize = transport_send_bin_finalize;
|
|
|
|
element_class->send_event = gst_transport_send_bin_element_event;
|
|
element_class->query = gst_transport_send_bin_element_query;
|
|
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_STREAM,
|
|
g_param_spec_object ("stream", "Stream",
|
|
"The TransportStream for this sending bin",
|
|
transport_stream_get_type (),
|
|
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
|
|
}
|
|
|
|
static void
|
|
transport_send_bin_init (TransportSendBin * send)
|
|
{
|
|
g_mutex_init (TSB_GET_LOCK (send));
|
|
}
|