gstreamer/ext/webrtc/transportsendbin.c
Olivier Crête ee0124cb36 webrtc: Remove the webrtc-priv.h header from public headers
And this time for real, also import it in a couple more places
inside the webrtc element to make it build.

Fixes #1607

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2359>
2021-06-28 16:06:59 +00:00

537 lines
16 KiB
C

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