gstreamer/ext/webrtc/transportsendbin.c

491 lines
17 KiB
C
Raw Normal View History

/* 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"
/*
* ,------------------------transport_send_%u-------------------------,
* ; ,-----dtlssrtpenc---, ;
* rtp_sink o--------------------------o rtp_sink_0 ; ,---nicesink---, ;
* ; ; src o--o sink ; ;
* ; ,--outputselector--, ,-o rtcp_sink_0 ; '--------------' ;
* ; ; src_0 o-' '-------------------' ;
* rtcp_sink ;---o sink ; ,----dtlssrtpenc----, ,---nicesink---, ;
* ; ; src_1 o---o rtcp_sink_0 src o--o sink ; ;
* ; '------------------' '-------------------' '--------------' ;
* '------------------------------------------------------------------'
*
* outputselecter is used to switch between rtcp-mux and no rtcp-mux
*
* 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"));
enum
{
PROP_0,
PROP_STREAM,
PROP_RTCP_MUX,
};
static void cleanup_blocks (TransportSendBin * send);
static void
_set_rtcp_mux (TransportSendBin * send, gboolean rtcp_mux)
{
GstPad *active_pad;
if (rtcp_mux)
active_pad = gst_element_get_static_pad (send->outputselector, "src_0");
else
active_pad = gst_element_get_static_pad (send->outputselector, "src_1");
send->rtcp_mux = rtcp_mux;
GST_OBJECT_UNLOCK (send);
g_object_set (send->outputselector, "active-pad", active_pad, NULL);
gst_object_unref (active_pad);
GST_OBJECT_LOCK (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? */
send->stream = TRANSPORT_STREAM (g_value_get_object (value));
break;
case PROP_RTCP_MUX:
_set_rtcp_mux (send, g_value_get_boolean (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;
case PROP_RTCP_MUX:
g_value_set_boolean (value, send->rtcp_mux);
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)
{
GST_LOG_OBJECT (pad, "blocking pad with data %" GST_PTR_FORMAT, info->data);
return GST_PAD_PROBE_OK;
}
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 */
gst_element_set_locked_state (send->stream->transport->dtlssrtpenc, TRUE);
gst_element_set_locked_state (send->stream->rtcp_transport->dtlssrtpenc,
TRUE);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:{
GstElement *elem;
GstPad *pad;
/* unblock the encoder once the key is set, this should also be automatic */
elem = send->stream->transport->dtlssrtpenc;
pad = gst_element_get_static_pad (elem, "rtp_sink_0");
send->rtp_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtp_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
/* unblock the encoder once the key is set, this should also be automatic */
pad = gst_element_get_static_pad (elem, "rtcp_sink_0");
send->rtcp_mux_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtcp_mux_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
elem = send->stream->rtcp_transport->dtlssrtpenc;
/* unblock the encoder once the key is set, this should also be automatic */
pad = gst_element_get_static_pad (elem, "rtcp_sink_0");
send->rtcp_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtcp_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
/* unblock ice sink once a connection is made, this should also be automatic */
elem = send->stream->transport->transport->sink;
pad = gst_element_get_static_pad (elem, "sink");
send->rtp_nice_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtp_nice_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
/* unblock ice sink once a connection is made, this should also be automatic */
elem = send->stream->rtcp_transport->transport->sink;
pad = gst_element_get_static_pad (elem, "sink");
send->rtcp_nice_block = _create_pad_block (elem, pad, 0, NULL, NULL);
send->rtcp_nice_block->block_id =
gst_pad_add_probe (pad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
GST_PAD_PROBE_TYPE_BUFFER_LIST, (GstPadProbeCallback) pad_block, NULL,
NULL);
gst_object_unref (pad);
break;
}
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
/* Normally, we do downward state change cleanups after the element
* has been stopped, as this will have set pads to flushing as needed
* and unblocked any pad probes that are blocked, but sometimes that's
* causing a deadlock on the build server in tests, with a race around
* the pad blocking/release timing, so free the pad blocks before
* stopping everything */
if (send->rtp_block && send->rtp_block->block_id) {
gst_pad_remove_probe (send->rtp_block->pad, send->rtp_block->block_id);
send->rtp_block->block_id = 0;
}
if (send->rtcp_mux_block && send->rtcp_mux_block->block_id) {
gst_pad_remove_probe (send->rtcp_mux_block->pad,
send->rtcp_mux_block->block_id);
send->rtcp_mux_block->block_id = 0;
}
if (send->rtcp_block && send->rtcp_block->block_id) {
gst_pad_remove_probe (send->rtcp_block->pad,
send->rtcp_block->block_id);
send->rtcp_block->block_id = 0;
}
if (send->rtp_nice_block && send->rtp_nice_block->block_id) {
gst_pad_remove_probe (send->rtp_nice_block->pad,
send->rtp_nice_block->block_id);
send->rtp_nice_block->block_id = 0;
}
if (send->rtcp_nice_block && send->rtcp_nice_block->block_id) {
gst_pad_remove_probe (send->rtcp_nice_block->pad,
send->rtcp_nice_block->block_id);
send->rtcp_nice_block->block_id = 0;
}
break;
}
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:{
GstElement *elem;
cleanup_blocks (send);
elem = send->stream->transport->dtlssrtpenc;
gst_element_set_locked_state (elem, FALSE);
elem = send->stream->rtcp_transport->dtlssrtpenc;
gst_element_set_locked_state (elem, FALSE);
break;
}
default:
break;
}
return ret;
}
static void
_on_dtls_enc_key_set (GstElement * element, TransportSendBin * send)
{
if (element == send->stream->transport->dtlssrtpenc) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtp_block->pad);
_free_pad_block (send->rtp_block);
send->rtp_block = NULL;
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtcp_mux_block->pad);
_free_pad_block (send->rtcp_mux_block);
send->rtcp_mux_block = NULL;
} else if (element == send->stream->rtcp_transport->dtlssrtpenc) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtcp_block->pad);
_free_pad_block (send->rtcp_block);
send->rtcp_block = NULL;
}
}
static void
_on_notify_ice_connection_state (GstWebRTCICETransport * transport,
GParamSpec * pspec, TransportSendBin * send)
{
GstWebRTCICEConnectionState state;
g_object_get (transport, "state", &state, NULL);
if (state == GST_WEBRTC_ICE_CONNECTION_STATE_CONNECTED ||
state == GST_WEBRTC_ICE_CONNECTION_STATE_COMPLETED) {
GST_OBJECT_LOCK (send);
if (transport == send->stream->transport->transport) {
if (send->rtp_nice_block) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtp_nice_block->pad);
_free_pad_block (send->rtp_nice_block);
}
send->rtp_nice_block = NULL;
} else if (transport == send->stream->rtcp_transport->transport) {
if (send->rtcp_nice_block) {
GST_LOG_OBJECT (send, "Unblocking pad %" GST_PTR_FORMAT,
send->rtcp_nice_block->pad);
_free_pad_block (send->rtcp_nice_block);
}
send->rtcp_nice_block = NULL;
}
GST_OBJECT_UNLOCK (send);
}
}
static void
transport_send_bin_constructed (GObject * object)
{
TransportSendBin *send = TRANSPORT_SEND_BIN (object);
GstWebRTCDTLSTransport *transport;
GstPadTemplate *templ;
GstPad *ghost, *pad;
g_return_if_fail (send->stream);
g_object_bind_property (send, "rtcp-mux", send->stream, "rtcp-mux",
G_BINDING_BIDIRECTIONAL);
transport = send->stream->transport;
templ = _find_pad_template (transport->dtlssrtpenc,
GST_PAD_SINK, GST_PAD_REQUEST, "rtp_sink_%d");
pad = gst_element_request_pad (transport->dtlssrtpenc, templ, "rtp_sink_0",
NULL);
/* unblock the encoder once the key is set */
g_signal_connect (transport->dtlssrtpenc, "on-key-set",
G_CALLBACK (_on_dtls_enc_key_set), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->dtlssrtpenc));
/* unblock ice sink once it signals a connection */
g_signal_connect (transport->transport, "notify::state",
G_CALLBACK (_on_notify_ice_connection_state), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->transport->sink));
if (!gst_element_link_pads (GST_ELEMENT (transport->dtlssrtpenc), "src",
GST_ELEMENT (transport->transport->sink), "sink"))
g_warn_if_reached ();
send->outputselector = gst_element_factory_make ("output-selector", NULL);
gst_bin_add (GST_BIN (send), send->outputselector);
if (!gst_element_link_pads (GST_ELEMENT (send->outputselector), "src_0",
GST_ELEMENT (transport->dtlssrtpenc), "rtcp_sink_0"))
g_warn_if_reached ();
ghost = gst_ghost_pad_new ("rtp_sink", pad);
gst_element_add_pad (GST_ELEMENT (send), ghost);
gst_object_unref (pad);
transport = send->stream->rtcp_transport;
templ = _find_pad_template (transport->dtlssrtpenc,
GST_PAD_SINK, GST_PAD_REQUEST, "rtcp_sink_%d");
/* unblock the encoder once the key is set */
g_signal_connect (transport->dtlssrtpenc, "on-key-set",
G_CALLBACK (_on_dtls_enc_key_set), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->dtlssrtpenc));
/* unblock ice sink once it signals a connection */
g_signal_connect (transport->transport, "notify::state",
G_CALLBACK (_on_notify_ice_connection_state), send);
gst_bin_add (GST_BIN (send), GST_ELEMENT (transport->transport->sink));
if (!gst_element_link_pads (GST_ELEMENT (transport->dtlssrtpenc), "src",
GST_ELEMENT (transport->transport->sink), "sink"))
g_warn_if_reached ();
if (!gst_element_link_pads (GST_ELEMENT (send->outputselector), "src_1",
GST_ELEMENT (transport->dtlssrtpenc), "rtcp_sink_0"))
g_warn_if_reached ();
pad = gst_element_get_static_pad (send->outputselector, "sink");
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_mux_block)
_free_pad_block (send->rtcp_mux_block);
send->rtcp_mux_block = NULL;
if (send->rtcp_block)
_free_pad_block (send->rtcp_block);
send->rtcp_block = NULL;
if (send->rtp_nice_block)
_free_pad_block (send->rtp_nice_block);
send->rtp_nice_block = NULL;
if (send->rtcp_nice_block)
_free_pad_block (send->rtcp_nice_block);
send->rtcp_nice_block = NULL;
}
static void
transport_send_bin_dispose (GObject * object)
{
TransportSendBin *send = TRANSPORT_SEND_BIN (object);
if (send->stream) {
g_signal_handlers_disconnect_by_data (send->stream->transport->transport,
send);
g_signal_handlers_disconnect_by_data (send->stream->
rtcp_transport->transport, send);
}
send->stream = NULL;
cleanup_blocks (send);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
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_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;
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));
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));
}
static void
transport_send_bin_init (TransportSendBin * send)
{
}