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gstreamer/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
Johan Sternerup 5b64cfaca3 webrtcice: Add webrtc ALPN header for HTTP proxy 
Section 3.4 in RFC8835 states that if a WebRTC endpoint uses an HTTP
proxy to access the Internet it MUST include the "ALPN" header. This
commit adds this header.

By default the ALPN used when connecting to the TURN/TCP server via a
proxy is set to "webrtc". It can be changed by adding an alpn url
option for the http-proxy. For example:

http://user:pass@my.http.proxy.com:8080?alpn=c-webrtc

This will add the header "ALPN: c-webrtc" to the HTTP proxy CONNECT
request.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4212>
2023-08-17 00:45:05 +00:00

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/* 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 "gstwebrtcbin.h"
#include "gstwebrtcstats.h"
#include "transportstream.h"
#include "transportreceivebin.h"
#include "utils.h"
#include "webrtcsdp.h"
#include "webrtctransceiver.h"
#include "webrtcdatachannel.h"
#include "webrtcsctptransport.h"
#include "gst/webrtc/webrtc-priv.h"
#include <gst/webrtc/nice/nice.h>
#include <gst/rtp/rtp.h>
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#define RANDOM_SESSION_ID \
((((((guint64) g_random_int()) << 32) | \
(guint64) g_random_int ())) & \
G_GUINT64_CONSTANT (0x7fffffffffffffff))
#define PC_GET_LOCK(w) (&w->priv->pc_lock)
#define PC_LOCK(w) (g_mutex_lock (PC_GET_LOCK(w)))
#define PC_UNLOCK(w) (g_mutex_unlock (PC_GET_LOCK(w)))
#define PC_GET_COND(w) (&w->priv->pc_cond)
#define PC_COND_WAIT(w) (g_cond_wait(PC_GET_COND(w), PC_GET_LOCK(w)))
#define PC_COND_BROADCAST(w) (g_cond_broadcast(PC_GET_COND(w)))
#define PC_COND_SIGNAL(w) (g_cond_signal(PC_GET_COND(w)))
#define ICE_GET_LOCK(w) (&w->priv->ice_lock)
#define ICE_LOCK(w) (g_mutex_lock (ICE_GET_LOCK(w)))
#define ICE_UNLOCK(w) (g_mutex_unlock (ICE_GET_LOCK(w)))
#define DC_GET_LOCK(w) (&w->priv->dc_lock)
#define DC_LOCK(w) (g_mutex_lock (DC_GET_LOCK(w)))
#define DC_UNLOCK(w) (g_mutex_unlock (DC_GET_LOCK(w)))
/* The extra time for the rtpstorage compared to the RTP jitterbuffer (in ms) */
#define RTPSTORAGE_EXTRA_TIME (50)
#define DEFAULT_JB_LATENCY 200
#define RTPHDREXT_MID GST_RTP_HDREXT_BASE "sdes:mid"
#define RTPHDREXT_STREAM_ID GST_RTP_HDREXT_BASE "sdes:rtp-stream-id"
#define RTPHDREXT_REPAIRED_STREAM_ID GST_RTP_HDREXT_BASE "sdes:repaired-rtp-stream-id"
#if !GLIB_CHECK_VERSION(2, 74, 0)
#define G_CONNECT_DEFAULT 0
#endif
/**
* SECTION: element-webrtcbin
* title: webrtcbin
*
* This webrtcbin implements the majority of the W3's peerconnection API and
* implementation guide where possible. Generating offers, answers and setting
* local and remote SDP's are all supported. Both media descriptions and
* descriptions involving data channels are supported.
*
* Each input/output pad is equivalent to a Track in W3 parlance which are
* added/removed from the bin. The number of requested sink pads is the number
* of streams that will be sent to the receiver and will be associated with a
* GstWebRTCRTPTransceiver (very similar to W3 RTPTransceiver's).
*
* On the receiving side, RTPTransceiver's are created in response to setting
* a remote description. Output pads for the receiving streams in the set
* description are also created when data is received.
*
* A TransportStream is created when needed in order to transport the data over
* the necessary DTLS/ICE channel to the peer. The exact configuration depends
* on the negotiated SDP's between the peers based on the bundle and rtcp
* configuration. Some cases are outlined below for a simple single
* audio/video/data session:
*
* - max-bundle uses a single transport for all
* media/data transported. Renegotiation involves adding/removing the
* necessary streams to the existing transports.
* - max-compat involves two TransportStream per media stream
* to transport the rtp and the rtcp packets and a single TransportStream for
* all data channels. Each stream change involves modifying the associated
* TransportStream/s as necessary.
*/
/*
* TODO:
* assert sending payload type matches the stream
* reconfiguration (of anything)
* LS groups
* balanced bundle policy
* setting custom DTLS certificates
*
* separate session id's from mlineindex properly
* how to deal with replacing a input/output track/stream
*/
static void _update_need_negotiation (GstWebRTCBin * webrtc);
static GstPad *_connect_input_stream (GstWebRTCBin * webrtc,
GstWebRTCBinPad * pad);
#define GST_CAT_DEFAULT gst_webrtc_bin_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp"));
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp"));
enum
{
PROP_PAD_TRANSCEIVER = 1,
};
static gboolean
_have_nice_elements (GstWebRTCBin * webrtc)
{
GstPluginFeature *feature;
feature = gst_registry_lookup_feature (gst_registry_get (), "nicesrc");
if (feature) {
gst_object_unref (feature);
} else {
GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
("%s", "libnice elements are not available"));
return FALSE;
}
feature = gst_registry_lookup_feature (gst_registry_get (), "nicesink");
if (feature) {
gst_object_unref (feature);
} else {
GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
("%s", "libnice elements are not available"));
return FALSE;
}
return TRUE;
}
static gboolean
_have_sctp_elements (GstWebRTCBin * webrtc)
{
GstPluginFeature *feature;
feature = gst_registry_lookup_feature (gst_registry_get (), "sctpdec");
if (feature) {
gst_object_unref (feature);
} else {
GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
("%s", "sctp elements are not available"));
return FALSE;
}
feature = gst_registry_lookup_feature (gst_registry_get (), "sctpenc");
if (feature) {
gst_object_unref (feature);
} else {
GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
("%s", "sctp elements are not available"));
return FALSE;
}
return TRUE;
}
static gboolean
_have_dtls_elements (GstWebRTCBin * webrtc)
{
GstPluginFeature *feature;
feature = gst_registry_lookup_feature (gst_registry_get (), "dtlsdec");
if (feature) {
gst_object_unref (feature);
} else {
GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
("%s", "dtls elements are not available"));
return FALSE;
}
feature = gst_registry_lookup_feature (gst_registry_get (), "dtlsenc");
if (feature) {
gst_object_unref (feature);
} else {
GST_ELEMENT_ERROR (webrtc, CORE, MISSING_PLUGIN, NULL,
("%s", "dtls elements are not available"));
return FALSE;
}
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
gst_webrtc_bin_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object);
switch (prop_id) {
case PROP_PAD_TRANSCEIVER:
g_value_set_object (value, pad->trans);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_bin_pad_finalize (GObject * object)
{
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object);
gst_clear_object (&pad->trans);
gst_clear_caps (&pad->received_caps);
g_clear_pointer (&pad->msid, g_free);
G_OBJECT_CLASS (gst_webrtc_bin_pad_parent_class)->finalize (object);
}
static void
gst_webrtc_bin_pad_class_init (GstWebRTCBinPadClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->get_property = gst_webrtc_bin_pad_get_property;
gobject_class->finalize = gst_webrtc_bin_pad_finalize;
g_object_class_install_property (gobject_class,
PROP_PAD_TRANSCEIVER,
g_param_spec_object ("transceiver", "Transceiver",
"Transceiver associated with this pad",
GST_TYPE_WEBRTC_RTP_TRANSCEIVER,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
}
static void
gst_webrtc_bin_pad_update_tos_event (GstWebRTCBinPad * wpad)
{
WebRTCTransceiver *trans = (WebRTCTransceiver *) wpad->trans;
if (wpad->received_caps && trans->parent.mid) {
GstPad *pad = GST_PAD (wpad);
gst_event_take (&trans->tos_event,
gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM_STICKY,
gst_structure_new ("GstWebRtcBinUpdateTos", "mid", G_TYPE_STRING,
trans->parent.mid, NULL)));
GST_DEBUG_OBJECT (pad, "sending new tos event %" GST_PTR_FORMAT,
trans->tos_event);
gst_pad_send_event (pad, gst_event_ref (trans->tos_event));
}
}
static GList *
_get_pending_sink_transceiver (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
GList *ret;
for (ret = webrtc->priv->pending_sink_transceivers; ret; ret = ret->next) {
if (ret->data == pad)
break;
}
return ret;
}
static gboolean
gst_webrtcbin_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (parent);
gboolean check_negotiation = FALSE;
if (GST_EVENT_TYPE (event) == GST_EVENT_CAPS) {
GstCaps *caps;
gst_event_parse_caps (event, &caps);
check_negotiation = (!wpad->received_caps
|| !gst_caps_is_equal (wpad->received_caps, caps));
gst_caps_replace (&wpad->received_caps, caps);
GST_DEBUG_OBJECT (parent,
"On %" GST_PTR_FORMAT " checking negotiation? %u, caps %"
GST_PTR_FORMAT, pad, check_negotiation, caps);
if (check_negotiation) {
gst_webrtc_bin_pad_update_tos_event (wpad);
}
/* A remote description might have been set while the pad hadn't
* yet received caps, delaying the connection of the input stream
*/
PC_LOCK (webrtc);
if (wpad->trans) {
GST_OBJECT_LOCK (wpad->trans);
if (wpad->trans->current_direction ==
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY
|| wpad->trans->current_direction ==
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GList *pending = _get_pending_sink_transceiver (webrtc, wpad);
if (pending) {
GST_LOG_OBJECT (pad, "Connecting input stream to rtpbin with "
"transceiver %" GST_PTR_FORMAT " and caps %" GST_PTR_FORMAT,
wpad->trans, wpad->received_caps);
_connect_input_stream (webrtc, wpad);
gst_pad_remove_probe (GST_PAD (pad), wpad->block_id);
wpad->block_id = 0;
gst_object_unref (pending->data);
webrtc->priv->pending_sink_transceivers =
g_list_delete_link (webrtc->priv->pending_sink_transceivers,
pending);
}
}
GST_OBJECT_UNLOCK (wpad->trans);
}
PC_UNLOCK (webrtc);
} else if (GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
check_negotiation = TRUE;
}
if (check_negotiation) {
PC_LOCK (webrtc);
_update_need_negotiation (webrtc);
PC_UNLOCK (webrtc);
}
return gst_pad_event_default (pad, parent, event);
}
static gboolean
gst_webrtcbin_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
GstWebRTCBinPad *wpad = GST_WEBRTC_BIN_PAD (pad);
gboolean ret = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_ACCEPT_CAPS:
GST_OBJECT_LOCK (wpad->trans);
if (wpad->trans->codec_preferences) {
GstCaps *caps;
gst_query_parse_accept_caps (query, &caps);
gst_query_set_accept_caps_result (query,
gst_caps_can_intersect (caps, wpad->trans->codec_preferences));
ret = TRUE;
}
GST_OBJECT_UNLOCK (wpad->trans);
break;
case GST_QUERY_CAPS:
{
GstCaps *codec_preferences = NULL;
GST_OBJECT_LOCK (wpad->trans);
if (wpad->trans->codec_preferences)
codec_preferences = gst_caps_ref (wpad->trans->codec_preferences);
GST_OBJECT_UNLOCK (wpad->trans);
if (codec_preferences) {
GstCaps *filter = NULL;
GstCaps *filter_prefs = NULL;
GstPad *target;
gst_query_parse_caps (query, &filter);
if (filter) {
filter_prefs = gst_caps_intersect_full (filter, codec_preferences,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (codec_preferences);
} else {
filter_prefs = codec_preferences;
}
target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
if (target) {
GstCaps *result;
result = gst_pad_query_caps (target, filter_prefs);
gst_query_set_caps_result (query, result);
gst_caps_unref (result);
gst_object_unref (target);
} else {
gst_query_set_caps_result (query, filter_prefs);
}
gst_caps_unref (filter_prefs);
ret = TRUE;
}
break;
}
default:
break;
}
if (ret)
return TRUE;
return gst_pad_query_default (pad, parent, query);
}
static void
gst_webrtc_bin_pad_init (GstWebRTCBinPad * pad)
{
}
static GstWebRTCBinPad *
gst_webrtc_bin_pad_new (const gchar * name, GstPadDirection direction,
char *msid)
{
GstWebRTCBinPad *pad;
GstPadTemplate *template;
GType pad_type;
if (direction == GST_PAD_SINK) {
template = gst_static_pad_template_get (&sink_template);
pad_type = GST_TYPE_WEBRTC_BIN_SINK_PAD;
} else if (direction == GST_PAD_SRC) {
template = gst_static_pad_template_get (&src_template);
pad_type = GST_TYPE_WEBRTC_BIN_SRC_PAD;
} else {
g_assert_not_reached ();
}
pad =
g_object_new (pad_type, "name", name, "direction",
direction, "template", template, NULL);
gst_object_unref (template);
pad->msid = msid;
GST_DEBUG_OBJECT (pad, "new visible pad with direction %s",
direction == GST_PAD_SRC ? "src" : "sink");
return pad;
}
enum
{
PROP_SINK_PAD_MSID = 1,
};
/**
* GstWebRTCBinSinkPad:
*
* Since: 1.22
*/
struct _GstWebRTCBinSinkPad
{
GstWebRTCBinPad pad;
};
G_DEFINE_TYPE (GstWebRTCBinSinkPad, gst_webrtc_bin_sink_pad,
GST_TYPE_WEBRTC_BIN_PAD);
static void
gst_webrtc_bin_sink_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object);
switch (prop_id) {
case PROP_SINK_PAD_MSID:
g_value_set_string (value, pad->msid);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_bin_sink_pad_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object);
switch (prop_id) {
case PROP_SINK_PAD_MSID:
g_free (pad->msid);
pad->msid = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_bin_sink_pad_class_init (GstWebRTCBinSinkPadClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->get_property = gst_webrtc_bin_sink_pad_get_property;
gobject_class->set_property = gst_webrtc_bin_sink_pad_set_property;
/**
* GstWebRTCBinSinkPad:msid:
*
* The MediaStream Identifier to use for this pad (MediaStreamTrack).
* Fallback is the RTP SDES cname value if not provided.
*
* Since: 1.22
*/
g_object_class_install_property (gobject_class,
PROP_SINK_PAD_MSID,
g_param_spec_string ("msid", "MSID",
"Local MediaStream ID to use for this pad (NULL = unset)", NULL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void
gst_webrtc_bin_sink_pad_init (GstWebRTCBinSinkPad * pad)
{
gst_pad_set_event_function (GST_PAD (pad), gst_webrtcbin_sink_event);
gst_pad_set_query_function (GST_PAD (pad), gst_webrtcbin_sink_query);
}
enum
{
PROP_SRC_PAD_MSID = 1,
};
/**
* GstWebRTCBinSrcPad:
*
* Since: 1.22
*/
struct _GstWebRTCBinSrcPad
{
GstWebRTCBinPad pad;
};
G_DEFINE_TYPE (GstWebRTCBinSrcPad, gst_webrtc_bin_src_pad,
GST_TYPE_WEBRTC_BIN_PAD);
static void
gst_webrtc_bin_src_pad_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (object);
switch (prop_id) {
case PROP_SRC_PAD_MSID:
g_value_set_string (value, pad->msid);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_bin_src_pad_class_init (GstWebRTCBinSrcPadClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->get_property = gst_webrtc_bin_src_pad_get_property;
/**
* GstWebRTCBinSrcPad:msid:
*
* The MediaStream Identifier the remote peer used for this pad (MediaStreamTrack).
* Will be NULL if not advertised in the remote SDP.
*
* Since: 1.22
*/
g_object_class_install_property (gobject_class,
PROP_SRC_PAD_MSID,
g_param_spec_string ("msid", "MSID",
"Remote MediaStream ID in use for this pad (NULL = not advertised)",
NULL, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
}
static void
gst_webrtc_bin_src_pad_init (GstWebRTCBinSrcPad * pad)
{
}
#define gst_webrtc_bin_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWebRTCBin, gst_webrtc_bin, GST_TYPE_BIN,
G_ADD_PRIVATE (GstWebRTCBin)
GST_DEBUG_CATEGORY_INIT (gst_webrtc_bin_debug, "webrtcbin", 0,
"webrtcbin element"););
enum
{
SIGNAL_0,
CREATE_OFFER_SIGNAL,
CREATE_ANSWER_SIGNAL,
SET_LOCAL_DESCRIPTION_SIGNAL,
SET_REMOTE_DESCRIPTION_SIGNAL,
ADD_ICE_CANDIDATE_SIGNAL,
ON_NEGOTIATION_NEEDED_SIGNAL,
ON_ICE_CANDIDATE_SIGNAL,
ON_NEW_TRANSCEIVER_SIGNAL,
GET_STATS_SIGNAL,
ADD_TRANSCEIVER_SIGNAL,
GET_TRANSCEIVER_SIGNAL,
GET_TRANSCEIVERS_SIGNAL,
ADD_TURN_SERVER_SIGNAL,
CREATE_DATA_CHANNEL_SIGNAL,
ON_DATA_CHANNEL_SIGNAL,
PREPARE_DATA_CHANNEL_SIGNAL,
REQUEST_AUX_SENDER,
ADD_ICE_CANDIDATE_FULL_SIGNAL,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_CONNECTION_STATE,
PROP_SIGNALING_STATE,
PROP_ICE_GATHERING_STATE,
PROP_ICE_CONNECTION_STATE,
PROP_LOCAL_DESCRIPTION,
PROP_CURRENT_LOCAL_DESCRIPTION,
PROP_PENDING_LOCAL_DESCRIPTION,
PROP_REMOTE_DESCRIPTION,
PROP_CURRENT_REMOTE_DESCRIPTION,
PROP_PENDING_REMOTE_DESCRIPTION,
PROP_STUN_SERVER,
PROP_TURN_SERVER,
PROP_BUNDLE_POLICY,
PROP_ICE_TRANSPORT_POLICY,
PROP_ICE_AGENT,
PROP_LATENCY,
PROP_SCTP_TRANSPORT,
PROP_HTTP_PROXY
};
static guint gst_webrtc_bin_signals[LAST_SIGNAL] = { 0 };
typedef struct
{
guint session_id;
GstWebRTCICEStream *stream;
} IceStreamItem;
/* FIXME: locking? */
GstWebRTCICEStream *
_find_ice_stream_for_session (GstWebRTCBin * webrtc, guint session_id)
{
int i;
for (i = 0; i < webrtc->priv->ice_stream_map->len; i++) {
IceStreamItem *item =
&g_array_index (webrtc->priv->ice_stream_map, IceStreamItem, i);
if (item->session_id == session_id) {
GST_TRACE_OBJECT (webrtc, "Found ice stream id %" GST_PTR_FORMAT " for "
"session %u", item->stream, session_id);
return item->stream;
}
}
GST_TRACE_OBJECT (webrtc, "No ice stream available for session %u",
session_id);
return NULL;
}
void
_add_ice_stream_item (GstWebRTCBin * webrtc, guint session_id,
GstWebRTCICEStream * stream)
{
IceStreamItem item = { session_id, stream };
GST_TRACE_OBJECT (webrtc, "adding ice stream %" GST_PTR_FORMAT " for "
"session %u", stream, session_id);
g_array_append_val (webrtc->priv->ice_stream_map, item);
}
typedef gboolean (*FindTransceiverFunc) (GstWebRTCRTPTransceiver * p1,
gconstpointer data);
static GstWebRTCRTPTransceiver *
_find_transceiver (GstWebRTCBin * webrtc, gconstpointer data,
FindTransceiverFunc func)
{
int i;
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *transceiver =
g_ptr_array_index (webrtc->priv->transceivers, i);
if (func (transceiver, data))
return transceiver;
}
return NULL;
}
static gboolean
transceiver_match_for_mid (GstWebRTCRTPTransceiver * trans, const gchar * mid)
{
return g_strcmp0 (trans->mid, mid) == 0;
}
static gboolean
transceiver_match_for_mline (GstWebRTCRTPTransceiver * trans, guint * mline)
{
if (trans->stopped)
return FALSE;
return trans->mline == *mline;
}
static GstWebRTCRTPTransceiver *
_find_transceiver_for_mline (GstWebRTCBin * webrtc, guint mlineindex)
{
GstWebRTCRTPTransceiver *trans;
trans = _find_transceiver (webrtc, &mlineindex,
(FindTransceiverFunc) transceiver_match_for_mline);
GST_TRACE_OBJECT (webrtc,
"Found transceiver %" GST_PTR_FORMAT " for mlineindex %u", trans,
mlineindex);
return trans;
}
static GstWebRTCRTPTransceiver *
_find_transceiver_for_mid (GstWebRTCBin * webrtc, const char *mid)
{
GstWebRTCRTPTransceiver *trans;
trans = _find_transceiver (webrtc, mid,
(FindTransceiverFunc) transceiver_match_for_mid);
GST_TRACE_OBJECT (webrtc, "Found transceiver %" GST_PTR_FORMAT " for "
"mid %s", trans, mid);
return trans;
}
typedef gboolean (*FindTransportFunc) (TransportStream * p1,
gconstpointer data);
static TransportStream *
_find_transport (GstWebRTCBin * webrtc, gconstpointer data,
FindTransportFunc func)
{
int i;
for (i = 0; i < webrtc->priv->transports->len; i++) {
TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i);
if (func (stream, data))
return stream;
}
return NULL;
}
static gboolean
match_stream_for_session (TransportStream * trans, guint * session)
{
return trans->session_id == *session;
}
static TransportStream *
_find_transport_for_session (GstWebRTCBin * webrtc, guint session_id)
{
TransportStream *stream;
stream = _find_transport (webrtc, &session_id,
(FindTransportFunc) match_stream_for_session);
GST_TRACE_OBJECT (webrtc,
"Found transport %" GST_PTR_FORMAT " for session %u", stream, session_id);
return stream;
}
static gboolean
match_stream_for_ice_transport (TransportStream * trans,
GstWebRTCICETransport * transport)
{
return trans->transport && trans->transport->transport == transport;
}
static TransportStream *
_find_transport_for_ice_transport (GstWebRTCBin * webrtc,
GstWebRTCICETransport * transport)
{
TransportStream *stream;
stream = _find_transport (webrtc, transport,
(FindTransportFunc) match_stream_for_ice_transport);
GST_TRACE_OBJECT (webrtc,
"Found transport %" GST_PTR_FORMAT " for ice transport %" GST_PTR_FORMAT,
stream, transport);
return stream;
}
typedef gboolean (*FindPadFunc) (GstWebRTCBinPad * p1, gconstpointer data);
static GstWebRTCBinPad *
_find_pad (GstWebRTCBin * webrtc, gconstpointer data, FindPadFunc func)
{
GstElement *element = GST_ELEMENT (webrtc);
GList *l;
GST_OBJECT_LOCK (webrtc);
l = element->pads;
for (; l; l = g_list_next (l)) {
if (!GST_IS_WEBRTC_BIN_PAD (l->data))
continue;
if (func (l->data, data)) {
gst_object_ref (l->data);
GST_OBJECT_UNLOCK (webrtc);
return l->data;
}
}
l = webrtc->priv->pending_pads;
for (; l; l = g_list_next (l)) {
if (!GST_IS_WEBRTC_BIN_PAD (l->data))
continue;
if (func (l->data, data)) {
gst_object_ref (l->data);
GST_OBJECT_UNLOCK (webrtc);
return l->data;
}
}
GST_OBJECT_UNLOCK (webrtc);
return NULL;
}
typedef gboolean (*FindDataChannelFunc) (WebRTCDataChannel * p1,
gconstpointer data);
static WebRTCDataChannel *
_find_data_channel (GstWebRTCBin * webrtc, gconstpointer data,
FindDataChannelFunc func)
{
int i;
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
WebRTCDataChannel *channel =
g_ptr_array_index (webrtc->priv->data_channels, i);
if (func (channel, data))
return channel;
}
return NULL;
}
static gboolean
data_channel_match_for_id (WebRTCDataChannel * channel, gint * id)
{
return channel->parent.id == *id;
}
/* always called with dc_lock held */
static WebRTCDataChannel *
_find_data_channel_for_id (GstWebRTCBin * webrtc, gint id)
{
WebRTCDataChannel *channel;
channel = _find_data_channel (webrtc, &id,
(FindDataChannelFunc) data_channel_match_for_id);
GST_TRACE_OBJECT (webrtc,
"Found data channel %" GST_PTR_FORMAT " for id %i", channel, id);
return channel;
}
static void
_add_pad_to_list (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
GST_OBJECT_LOCK (webrtc);
webrtc->priv->pending_pads = g_list_prepend (webrtc->priv->pending_pads, pad);
GST_OBJECT_UNLOCK (webrtc);
}
static gboolean
_remove_pending_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
gboolean ret = FALSE;
GList *l;
GST_OBJECT_LOCK (webrtc);
l = g_list_find (webrtc->priv->pending_pads, pad);
if (l) {
webrtc->priv->pending_pads =
g_list_remove_link (webrtc->priv->pending_pads, l);
g_list_free (l);
ret = TRUE;
}
GST_OBJECT_UNLOCK (webrtc);
return ret;
}
static void
_add_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
_remove_pending_pad (webrtc, pad);
if (webrtc->priv->running)
gst_pad_set_active (GST_PAD (pad), TRUE);
gst_element_add_pad (GST_ELEMENT (webrtc), GST_PAD (pad));
}
static void
_remove_pad (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
_remove_pending_pad (webrtc, pad);
gst_element_remove_pad (GST_ELEMENT (webrtc), GST_PAD (pad));
}
typedef struct
{
GstPadDirection direction;
guint mline;
} MLineMatch;
static gboolean
pad_match_for_mline (GstWebRTCBinPad * pad, const MLineMatch * match)
{
return GST_PAD_DIRECTION (pad) == match->direction
&& pad->trans->mline == match->mline;
}
static GstWebRTCBinPad *
_find_pad_for_mline (GstWebRTCBin * webrtc, GstPadDirection direction,
guint mline)
{
MLineMatch m = { direction, mline };
return _find_pad (webrtc, &m, (FindPadFunc) pad_match_for_mline);
}
typedef struct
{
GstPadDirection direction;
GstWebRTCRTPTransceiver *trans;
} TransMatch;
static gboolean
pad_match_for_transceiver (GstWebRTCBinPad * pad, TransMatch * m)
{
return GST_PAD_DIRECTION (pad) == m->direction && pad->trans == m->trans;
}
static GstWebRTCBinPad *
_find_pad_for_transceiver (GstWebRTCBin * webrtc, GstPadDirection direction,
GstWebRTCRTPTransceiver * trans)
{
TransMatch m = { direction, trans };
return _find_pad (webrtc, &m, (FindPadFunc) pad_match_for_transceiver);
}
#if 0
static gboolean
match_for_pad (GstWebRTCBinPad * pad, GstWebRTCBinPad * other)
{
return pad == other;
}
#endif
struct SsrcMatch
{
GstWebRTCRTPTransceiverDirection direction;
guint32 ssrc;
};
static gboolean
mid_ssrc_match_for_ssrc (SsrcMapItem * entry, const struct SsrcMatch *match)
{
return entry->direction == match->direction && entry->ssrc == match->ssrc;
}
static gboolean
mid_ssrc_remove_ssrc (SsrcMapItem * item, const struct SsrcMatch *match)
{
return !mid_ssrc_match_for_ssrc (item, match);
}
static SsrcMapItem *
find_mid_ssrc_for_ssrc (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiverDirection direction, guint rtp_session, guint ssrc)
{
TransportStream *stream = _find_transport_for_session (webrtc, rtp_session);
struct SsrcMatch m = { direction, ssrc };
if (!stream)
return NULL;
return transport_stream_find_ssrc_map_item (stream, &m,
(FindSsrcMapFunc) mid_ssrc_match_for_ssrc);
}
static SsrcMapItem *
find_or_add_ssrc_map_item (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiverDirection direction, guint rtp_session, guint ssrc,
guint media_idx)
{
TransportStream *stream = _find_transport_for_session (webrtc, rtp_session);
struct SsrcMatch m = { direction, ssrc };
SsrcMapItem *item;
if (!stream)
return NULL;
if ((item = transport_stream_find_ssrc_map_item (stream, &m,
(FindSsrcMapFunc) mid_ssrc_match_for_ssrc)))
return item;
return transport_stream_add_ssrc_map_item (stream, direction, ssrc,
media_idx);
}
static void
remove_ssrc_entry_by_ssrc (GstWebRTCBin * webrtc, guint rtp_session, guint ssrc)
{
TransportStream *stream;
stream = _find_transport_for_session (webrtc, rtp_session);
if (stream) {
struct SsrcMatch m =
{ GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, ssrc };
transport_stream_filter_ssrc_map_item (stream, &m,
(FindSsrcMapFunc) mid_ssrc_remove_ssrc);
m.direction = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY;
transport_stream_filter_ssrc_map_item (stream, &m,
(FindSsrcMapFunc) mid_ssrc_remove_ssrc);
}
}
static gboolean
_unlock_pc_thread (GMutex * lock)
{
g_mutex_unlock (lock);
return G_SOURCE_REMOVE;
}
static gpointer
_gst_pc_thread (GstWebRTCBin * webrtc)
{
PC_LOCK (webrtc);
webrtc->priv->main_context = g_main_context_new ();
webrtc->priv->loop = g_main_loop_new (webrtc->priv->main_context, FALSE);
PC_COND_BROADCAST (webrtc);
g_main_context_invoke (webrtc->priv->main_context,
(GSourceFunc) _unlock_pc_thread, PC_GET_LOCK (webrtc));
/* Having the thread be the thread default GMainContext will break the
* required queue-like ordering (from W3's peerconnection spec) of re-entrant
* tasks */
g_main_loop_run (webrtc->priv->loop);
GST_OBJECT_LOCK (webrtc);
g_main_context_unref (webrtc->priv->main_context);
webrtc->priv->main_context = NULL;
GST_OBJECT_UNLOCK (webrtc);
PC_LOCK (webrtc);
g_main_loop_unref (webrtc->priv->loop);
webrtc->priv->loop = NULL;
PC_COND_BROADCAST (webrtc);
PC_UNLOCK (webrtc);
return NULL;
}
static void
_start_thread (GstWebRTCBin * webrtc)
{
gchar *name;
PC_LOCK (webrtc);
name = g_strdup_printf ("%s:pc", GST_OBJECT_NAME (webrtc));
webrtc->priv->thread = g_thread_new (name, (GThreadFunc) _gst_pc_thread,
webrtc);
g_free (name);
while (!webrtc->priv->loop)
PC_COND_WAIT (webrtc);
webrtc->priv->is_closed = FALSE;
PC_UNLOCK (webrtc);
}
static void
_stop_thread (GstWebRTCBin * webrtc)
{
GST_OBJECT_LOCK (webrtc);
webrtc->priv->is_closed = TRUE;
GST_OBJECT_UNLOCK (webrtc);
PC_LOCK (webrtc);
g_main_loop_quit (webrtc->priv->loop);
while (webrtc->priv->loop)
PC_COND_WAIT (webrtc);
PC_UNLOCK (webrtc);
g_thread_unref (webrtc->priv->thread);
}
static gboolean
_execute_op (GstWebRTCBinTask * op)
{
GstStructure *s;
PC_LOCK (op->webrtc);
if (op->webrtc->priv->is_closed) {
PC_UNLOCK (op->webrtc);
if (op->promise) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"webrtcbin is closed. aborting execution.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
gst_promise_reply (op->promise, s);
g_clear_error (&error);
}
GST_DEBUG_OBJECT (op->webrtc,
"Peerconnection is closed, aborting execution");
goto out;
}
s = op->op (op->webrtc, op->data);
PC_UNLOCK (op->webrtc);
if (op->promise)
gst_promise_reply (op->promise, s);
else if (s)
gst_structure_free (s);
out:
return G_SOURCE_REMOVE;
}
static void
_free_op (GstWebRTCBinTask * op)
{
if (op->notify)
op->notify (op->data);
if (op->promise)
gst_promise_unref (op->promise);
g_free (op);
}
/*
* @promise is for correctly signalling the failure case to the caller when
* the user supplies it. Without passing it in, the promise would never
* be replied to in the case that @webrtc becomes closed between the idle
* source addition and the the execution of the idle source.
*/
gboolean
gst_webrtc_bin_enqueue_task (GstWebRTCBin * webrtc, GstWebRTCBinFunc func,
gpointer data, GDestroyNotify notify, GstPromise * promise)
{
GstWebRTCBinTask *op;
GMainContext *ctx;
GSource *source;
g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE);
GST_OBJECT_LOCK (webrtc);
if (webrtc->priv->is_closed) {
GST_OBJECT_UNLOCK (webrtc);
GST_DEBUG_OBJECT (webrtc, "Peerconnection is closed, aborting execution");
if (notify)
notify (data);
return FALSE;
}
ctx = g_main_context_ref (webrtc->priv->main_context);
GST_OBJECT_UNLOCK (webrtc);
op = g_new0 (GstWebRTCBinTask, 1);
op->webrtc = webrtc;
op->op = func;
op->data = data;
op->notify = notify;
if (promise)
op->promise = gst_promise_ref (promise);
source = g_idle_source_new ();
g_source_set_priority (source, G_PRIORITY_DEFAULT);
g_source_set_callback (source, (GSourceFunc) _execute_op, op,
(GDestroyNotify) _free_op);
g_source_attach (source, ctx);
g_source_unref (source);
g_main_context_unref (ctx);
return TRUE;
}
void
gst_webrtc_bin_get_peer_connection_stats (GstWebRTCBin * webrtc,
guint * data_channels_opened, guint * data_channels_closed)
{
DC_LOCK (webrtc);
if (data_channels_opened) {
*data_channels_opened = webrtc->priv->data_channels_opened;
}
if (data_channels_closed) {
*data_channels_closed = webrtc->priv->data_channels_closed;
}
DC_UNLOCK (webrtc);
}
/* https://www.w3.org/TR/webrtc/#dom-rtciceconnectionstate */
static GstWebRTCICEConnectionState
_collate_ice_connection_states (GstWebRTCBin * webrtc)
{
#define STATE(val) GST_WEBRTC_ICE_CONNECTION_STATE_ ## val
GstWebRTCICEConnectionState any_state = 0;
gboolean all_new_or_closed = TRUE;
gboolean all_completed_or_closed = TRUE;
gboolean all_connected_completed_or_closed = TRUE;
int i;
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *rtp_trans =
g_ptr_array_index (webrtc->priv->transceivers, i);
GstWebRTCICETransport *transport;
GstWebRTCICEConnectionState ice_state;
if (rtp_trans->stopped) {
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
continue;
}
if (!rtp_trans->mid) {
GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
continue;
}
transport = webrtc_transceiver_get_dtls_transport (rtp_trans)->transport;
/* get transport state */
g_object_get (transport, "state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p state 0x%x", rtp_trans,
ice_state);
any_state |= (1 << ice_state);
if (ice_state != STATE (NEW) && ice_state != STATE (CLOSED))
all_new_or_closed = FALSE;
if (ice_state != STATE (COMPLETED) && ice_state != STATE (CLOSED))
all_completed_or_closed = FALSE;
if (ice_state != STATE (CONNECTED) && ice_state != STATE (COMPLETED)
&& ice_state != STATE (CLOSED))
all_connected_completed_or_closed = FALSE;
}
GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x", any_state);
if (webrtc->priv->is_closed) {
GST_TRACE_OBJECT (webrtc, "returning closed");
return STATE (CLOSED);
}
/* Any of the RTCIceTransports are in the failed state. */
if (any_state & (1 << STATE (FAILED))) {
GST_TRACE_OBJECT (webrtc, "returning failed");
return STATE (FAILED);
}
/* Any of the RTCIceTransports are in the disconnected state. */
if (any_state & (1 << STATE (DISCONNECTED))) {
GST_TRACE_OBJECT (webrtc, "returning disconnected");
return STATE (DISCONNECTED);
}
/* All of the RTCIceTransports are in the new or closed state, or there are
* no transports. */
if (all_new_or_closed || webrtc->priv->transceivers->len == 0) {
GST_TRACE_OBJECT (webrtc, "returning new");
return STATE (NEW);
}
/* Any of the RTCIceTransports are in the checking or new state. */
if ((any_state & (1 << STATE (CHECKING))) || (any_state & (1 << STATE (NEW)))) {
GST_TRACE_OBJECT (webrtc, "returning checking");
return STATE (CHECKING);
}
/* All RTCIceTransports are in the completed or closed state. */
if (all_completed_or_closed) {
GST_TRACE_OBJECT (webrtc, "returning completed");
return STATE (COMPLETED);
}
/* All RTCIceTransports are in the connected, completed or closed state. */
if (all_connected_completed_or_closed) {
GST_TRACE_OBJECT (webrtc, "returning connected");
return STATE (CONNECTED);
}
GST_FIXME ("unspecified situation, returning old state");
return webrtc->ice_connection_state;
#undef STATE
}
/* https://www.w3.org/TR/webrtc/#dom-rtcicegatheringstate */
static GstWebRTCICEGatheringState
_collate_ice_gathering_states (GstWebRTCBin * webrtc)
{
#define STATE(val) GST_WEBRTC_ICE_GATHERING_STATE_ ## val
GstWebRTCICEGatheringState any_state = 0;
GstWebRTCICEGatheringState ice_state;
GstWebRTCDTLSTransport *dtls_transport;
GstWebRTCICETransport *transport;
gboolean all_completed = webrtc->priv->transceivers->len > 0 ||
webrtc->priv->data_channel_transport;
int i;
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *rtp_trans =
g_ptr_array_index (webrtc->priv->transceivers, i);
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
TransportStream *stream = trans->stream;
if (rtp_trans->stopped || stream == NULL) {
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped or unassociated",
rtp_trans);
continue;
}
/* We only have a mid in the transceiver after we got the SDP answer,
* which is usually long after gathering has finished */
if (!rtp_trans->mid) {
GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
}
dtls_transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
if (dtls_transport == NULL) {
GST_WARNING ("Transceiver %p has no DTLS transport", rtp_trans);
continue;
}
transport = dtls_transport->transport;
/* get gathering state */
g_object_get (transport, "gathering-state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p gathering state: 0x%x", rtp_trans,
ice_state);
any_state |= (1 << ice_state);
if (ice_state != STATE (COMPLETE))
all_completed = FALSE;
}
/* check data channel transport gathering state */
if (all_completed && webrtc->priv->data_channel_transport) {
if ((dtls_transport = webrtc->priv->data_channel_transport->transport)) {
transport = dtls_transport->transport;
g_object_get (transport, "gathering-state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc,
"data channel transport %p gathering state: 0x%x", dtls_transport,
ice_state);
any_state |= (1 << ice_state);
if (ice_state != STATE (COMPLETE))
all_completed = FALSE;
}
}
GST_TRACE_OBJECT (webrtc, "ICE gathering state: 0x%x", any_state);
/* Any of the RTCIceTransport s are in the gathering state. */
if (any_state & (1 << STATE (GATHERING))) {
GST_TRACE_OBJECT (webrtc, "returning gathering");
return STATE (GATHERING);
}
/* At least one RTCIceTransport exists, and all RTCIceTransport s are in
* the completed gathering state. */
if (all_completed) {
GST_TRACE_OBJECT (webrtc, "returning complete");
return STATE (COMPLETE);
}
/* Any of the RTCIceTransport s are in the new gathering state and none
* of the transports are in the gathering state, or there are no transports. */
GST_TRACE_OBJECT (webrtc, "returning new");
return STATE (NEW);
#undef STATE
}
/* https://www.w3.org/TR/webrtc/#rtcpeerconnectionstate-enum */
static GstWebRTCPeerConnectionState
_collate_peer_connection_states (GstWebRTCBin * webrtc)
{
#define STATE(v) GST_WEBRTC_PEER_CONNECTION_STATE_ ## v
#define ICE_STATE(v) GST_WEBRTC_ICE_CONNECTION_STATE_ ## v
#define DTLS_STATE(v) GST_WEBRTC_DTLS_TRANSPORT_STATE_ ## v
GstWebRTCICEConnectionState any_ice_state = 0;
GstWebRTCDTLSTransportState any_dtls_state = 0;
gboolean ice_all_new_or_closed = TRUE;
gboolean dtls_all_new_or_closed = TRUE;
gboolean ice_all_new_connecting_or_checking = TRUE;
gboolean dtls_all_new_connecting_or_checking = TRUE;
gboolean ice_all_connected_completed_or_closed = TRUE;
gboolean dtls_all_connected_completed_or_closed = TRUE;
int i;
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *rtp_trans =
g_ptr_array_index (webrtc->priv->transceivers, i);
GstWebRTCDTLSTransport *transport;
GstWebRTCICEConnectionState ice_state;
GstWebRTCDTLSTransportState dtls_state;
if (rtp_trans->stopped) {
GST_TRACE_OBJECT (webrtc, "transceiver %p stopped", rtp_trans);
continue;
}
if (!rtp_trans->mid) {
GST_TRACE_OBJECT (webrtc, "transceiver %p has no mid", rtp_trans);
continue;
}
transport = webrtc_transceiver_get_dtls_transport (rtp_trans);
/* get transport state */
g_object_get (transport, "state", &dtls_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p DTLS state: 0x%x", rtp_trans,
dtls_state);
any_dtls_state |= (1 << dtls_state);
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CLOSED))
dtls_all_new_or_closed = FALSE;
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CONNECTING))
dtls_all_new_connecting_or_checking = FALSE;
if (dtls_state != DTLS_STATE (CONNECTED)
&& dtls_state != DTLS_STATE (CLOSED))
dtls_all_connected_completed_or_closed = FALSE;
g_object_get (transport->transport, "state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc, "transceiver %p ICE state: 0x%x", rtp_trans,
ice_state);
any_ice_state |= (1 << ice_state);
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CLOSED))
ice_all_new_or_closed = FALSE;
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CHECKING))
ice_all_new_connecting_or_checking = FALSE;
if (ice_state != ICE_STATE (CONNECTED) && ice_state != ICE_STATE (COMPLETED)
&& ice_state != ICE_STATE (CLOSED))
ice_all_connected_completed_or_closed = FALSE;
}
// also check data channel transport state
if (webrtc->priv->data_channel_transport) {
GstWebRTCDTLSTransport *transport =
webrtc->priv->data_channel_transport->transport;
GstWebRTCICEConnectionState ice_state;
GstWebRTCDTLSTransportState dtls_state;
g_object_get (transport, "state", &dtls_state, NULL);
GST_TRACE_OBJECT (webrtc, "data channel transport DTLS state: 0x%x",
dtls_state);
any_dtls_state |= (1 << dtls_state);
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CLOSED))
dtls_all_new_or_closed = FALSE;
if (dtls_state != DTLS_STATE (NEW) && dtls_state != DTLS_STATE (CONNECTING))
dtls_all_new_connecting_or_checking = FALSE;
if (dtls_state != DTLS_STATE (CONNECTED)
&& dtls_state != DTLS_STATE (CLOSED))
dtls_all_connected_completed_or_closed = FALSE;
g_object_get (transport->transport, "state", &ice_state, NULL);
GST_TRACE_OBJECT (webrtc, "data channel transport ICE state: 0x%x",
ice_state);
any_ice_state |= (1 << ice_state);
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CLOSED))
ice_all_new_or_closed = FALSE;
if (ice_state != ICE_STATE (NEW) && ice_state != ICE_STATE (CHECKING))
ice_all_new_connecting_or_checking = FALSE;
if (ice_state != ICE_STATE (CONNECTED) && ice_state != ICE_STATE (COMPLETED)
&& ice_state != ICE_STATE (CLOSED))
ice_all_connected_completed_or_closed = FALSE;
}
GST_TRACE_OBJECT (webrtc, "ICE connection state: 0x%x. DTLS connection "
"state: 0x%x", any_ice_state, any_dtls_state);
/* The RTCPeerConnection object's [[ isClosed]] slot is true. */
if (webrtc->priv->is_closed) {
GST_TRACE_OBJECT (webrtc, "returning closed");
return STATE (CLOSED);
}
/* Any of the RTCIceTransport s or RTCDtlsTransport s are in a failed state. */
if (any_ice_state & (1 << ICE_STATE (FAILED))) {
GST_TRACE_OBJECT (webrtc, "returning failed");
return STATE (FAILED);
}
if (any_dtls_state & (1 << DTLS_STATE (FAILED))) {
GST_TRACE_OBJECT (webrtc, "returning failed");
return STATE (FAILED);
}
/* Any of the RTCIceTransport's or RTCDtlsTransport's are in the disconnected
* state. */
if (any_ice_state & (1 << ICE_STATE (DISCONNECTED))) {
GST_TRACE_OBJECT (webrtc, "returning disconnected");
return STATE (DISCONNECTED);
}
/* All RTCIceTransports and RTCDtlsTransports are in the new or closed
* state, or there are no transports. */
if ((dtls_all_new_or_closed && ice_all_new_or_closed)
|| webrtc->priv->transports->len == 0) {
GST_TRACE_OBJECT (webrtc, "returning new");
return STATE (NEW);
}
/* All RTCIceTransports and RTCDtlsTransports are in the new, connecting
* or checking state. */
if (dtls_all_new_connecting_or_checking && ice_all_new_connecting_or_checking) {
GST_TRACE_OBJECT (webrtc, "returning connecting");
return STATE (CONNECTING);
}
/* All RTCIceTransports and RTCDtlsTransports are in the connected,
* completed or closed state. */
if (dtls_all_connected_completed_or_closed
&& ice_all_connected_completed_or_closed) {
GST_TRACE_OBJECT (webrtc, "returning connected");
return STATE (CONNECTED);
}
/* FIXME: Unspecified state that happens for us */
if ((dtls_all_new_connecting_or_checking
|| dtls_all_connected_completed_or_closed)
&& (ice_all_new_connecting_or_checking
|| ice_all_connected_completed_or_closed)) {
GST_TRACE_OBJECT (webrtc, "returning connecting");
return STATE (CONNECTING);
}
GST_FIXME_OBJECT (webrtc,
"Undefined situation detected, returning old state");
return webrtc->peer_connection_state;
#undef DTLS_STATE
#undef ICE_STATE
#undef STATE
}
static GstStructure *
_update_ice_gathering_state_task (GstWebRTCBin * webrtc, gpointer data)
{
GstWebRTCICEGatheringState old_state = webrtc->ice_gathering_state;
GstWebRTCICEGatheringState new_state;
new_state = _collate_ice_gathering_states (webrtc);
/* If the new state is complete, before we update the public state,
* check if anyone published more ICE candidates while we were collating
* and stop if so, because it means there's a new later
* ice_gathering_state_task queued */
if (new_state == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) {
ICE_LOCK (webrtc);
if (webrtc->priv->pending_local_ice_candidates->len != 0) {
/* ICE candidates queued for emissiong -> we're gathering, not complete */
new_state = GST_WEBRTC_ICE_GATHERING_STATE_GATHERING;
}
ICE_UNLOCK (webrtc);
}
if (new_state != webrtc->ice_gathering_state) {
const gchar *old_s, *new_s;
old_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_GATHERING_STATE,
old_state);
new_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_GATHERING_STATE,
new_state);
GST_INFO_OBJECT (webrtc, "ICE gathering state change from %s(%u) to %s(%u)",
old_s, old_state, new_s, new_state);
webrtc->ice_gathering_state = new_state;
PC_UNLOCK (webrtc);
g_object_notify (G_OBJECT (webrtc), "ice-gathering-state");
PC_LOCK (webrtc);
}
return NULL;
}
static GstStructure *
_update_ice_connection_state_task (GstWebRTCBin * webrtc, gpointer data)
{
GstWebRTCICEConnectionState old_state = webrtc->ice_connection_state;
GstWebRTCICEConnectionState new_state;
new_state = _collate_ice_connection_states (webrtc);
if (new_state != old_state) {
const gchar *old_s, *new_s;
old_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_CONNECTION_STATE,
old_state);
new_s = _enum_value_to_string (GST_TYPE_WEBRTC_ICE_CONNECTION_STATE,
new_state);
GST_INFO_OBJECT (webrtc,
"ICE connection state change from %s(%u) to %s(%u)", old_s, old_state,
new_s, new_state);
webrtc->ice_connection_state = new_state;
PC_UNLOCK (webrtc);
g_object_notify (G_OBJECT (webrtc), "ice-connection-state");
PC_LOCK (webrtc);
}
return NULL;
}
static void
_update_ice_connection_state (GstWebRTCBin * webrtc)
{
gst_webrtc_bin_enqueue_task (webrtc, _update_ice_connection_state_task, NULL,
NULL, NULL);
}
static GstStructure *
_update_peer_connection_state_task (GstWebRTCBin * webrtc, gpointer data)
{
GstWebRTCPeerConnectionState old_state = webrtc->peer_connection_state;
GstWebRTCPeerConnectionState new_state;
new_state = _collate_peer_connection_states (webrtc);
if (new_state != old_state) {
const gchar *old_s, *new_s;
old_s = _enum_value_to_string (GST_TYPE_WEBRTC_PEER_CONNECTION_STATE,
old_state);
new_s = _enum_value_to_string (GST_TYPE_WEBRTC_PEER_CONNECTION_STATE,
new_state);
GST_INFO_OBJECT (webrtc,
"Peer connection state change from %s(%u) to %s(%u)", old_s, old_state,
new_s, new_state);
webrtc->peer_connection_state = new_state;
PC_UNLOCK (webrtc);
g_object_notify (G_OBJECT (webrtc), "connection-state");
PC_LOCK (webrtc);
}
return NULL;
}
static void
_update_peer_connection_state (GstWebRTCBin * webrtc)
{
gst_webrtc_bin_enqueue_task (webrtc, _update_peer_connection_state_task,
NULL, NULL, NULL);
}
static gboolean
_all_sinks_have_caps (GstWebRTCBin * webrtc)
{
GList *l;
gboolean res = FALSE;
GST_OBJECT_LOCK (webrtc);
l = GST_ELEMENT (webrtc)->pads;
for (; l; l = g_list_next (l)) {
GstWebRTCBinPad *wpad;
if (!GST_IS_WEBRTC_BIN_PAD (l->data))
continue;
wpad = GST_WEBRTC_BIN_PAD (l->data);
if (GST_PAD_DIRECTION (l->data) == GST_PAD_SINK && !wpad->received_caps
&& (!wpad->trans || !wpad->trans->stopped)) {
if (wpad->trans && wpad->trans->codec_preferences) {
continue;
} else {
goto done;
}
}
}
l = webrtc->priv->pending_pads;
for (; l; l = g_list_next (l)) {
if (!GST_IS_WEBRTC_BIN_PAD (l->data)) {
goto done;
}
}
res = TRUE;
done:
GST_OBJECT_UNLOCK (webrtc);
return res;
}
/* http://w3c.github.io/webrtc-pc/#dfn-check-if-negotiation-is-needed */
static gboolean
_check_if_negotiation_is_needed (GstWebRTCBin * webrtc)
{
int i;
GST_LOG_OBJECT (webrtc, "checking if negotiation is needed");
/* We can't negotiate until we have received caps on all our sink pads,
* as we will need the formats in our offer / answer */
if (!_all_sinks_have_caps (webrtc)) {
GST_LOG_OBJECT (webrtc,
"no negotiation possible until caps have been received on all sink pads");
return FALSE;
}
/* If any implementation-specific negotiation is required, as described at
* the start of this section, return "true".
* FIXME */
/* FIXME: emit when input caps/format changes? */
if (!webrtc->current_local_description) {
GST_LOG_OBJECT (webrtc, "no local description set");
return TRUE;
}
if (!webrtc->current_remote_description) {
GST_LOG_OBJECT (webrtc, "no remote description set");
return TRUE;
}
/* If connection has created any RTCDataChannel's, and no m= section has
* been negotiated yet for data, return "true". */
if (webrtc->priv->data_channels->len > 0) {
if (_message_get_datachannel_index (webrtc->current_local_description->
sdp) >= G_MAXUINT) {
GST_LOG_OBJECT (webrtc,
"no data channel media section and have %u " "transports",
webrtc->priv->data_channels->len);
return TRUE;
}
}
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *trans;
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
if (trans->stopped) {
/* FIXME: If t is stopped and is associated with an m= section according to
* [JSEP] (section 3.4.1.), but the associated m= section is not yet
* rejected in connection's currentLocalDescription or
* currentRemoteDescription , return "true". */
GST_FIXME_OBJECT (webrtc,
"check if the transceiver is rejected in descriptions");
} else {
const GstSDPMedia *media;
GstWebRTCRTPTransceiverDirection local_dir, remote_dir;
if (trans->mline == -1 || trans->mid == NULL) {
GST_LOG_OBJECT (webrtc, "unassociated transceiver %i %" GST_PTR_FORMAT
" mid %s", i, trans, trans->mid);
return TRUE;
}
/* internal inconsistency */
g_assert (trans->mline <
gst_sdp_message_medias_len (webrtc->current_local_description->sdp));
g_assert (trans->mline <
gst_sdp_message_medias_len (webrtc->current_remote_description->sdp));
/* FIXME: msid handling
* If t's direction is "sendrecv" or "sendonly", and the associated m=
* section in connection's currentLocalDescription doesn't contain an
* "a=msid" line, return "true". */
media =
gst_sdp_message_get_media (webrtc->current_local_description->sdp,
trans->mline);
local_dir = _get_direction_from_media (media);
media =
gst_sdp_message_get_media (webrtc->current_remote_description->sdp,
trans->mline);
remote_dir = _get_direction_from_media (media);
if (webrtc->current_local_description->type == GST_WEBRTC_SDP_TYPE_OFFER) {
/* If connection's currentLocalDescription if of type "offer", and
* the direction of the associated m= section in neither the offer
* nor answer matches t's direction, return "true". */
if (local_dir != trans->direction && remote_dir != trans->direction) {
GST_LOG_OBJECT (webrtc, "transceiver direction (%s) doesn't match "
"description (local %s remote %s)",
gst_webrtc_rtp_transceiver_direction_to_string (trans->direction),
gst_webrtc_rtp_transceiver_direction_to_string (local_dir),
gst_webrtc_rtp_transceiver_direction_to_string (remote_dir));
return TRUE;
}
} else if (webrtc->current_local_description->type ==
GST_WEBRTC_SDP_TYPE_ANSWER) {
GstWebRTCRTPTransceiverDirection intersect_dir;
/* If connection's currentLocalDescription if of type "answer", and
* the direction of the associated m= section in the answer does not
* match t's direction intersected with the offered direction (as
* described in [JSEP] (section 5.3.1.)), return "true". */
/* remote is the offer, local is the answer */
intersect_dir = _intersect_answer_directions (remote_dir, local_dir);
if (intersect_dir != trans->direction) {
GST_LOG_OBJECT (webrtc, "transceiver direction (%s) doesn't match "
"description intersected direction %s (local %s remote %s)",
gst_webrtc_rtp_transceiver_direction_to_string (trans->direction),
gst_webrtc_rtp_transceiver_direction_to_string (local_dir),
gst_webrtc_rtp_transceiver_direction_to_string (intersect_dir),
gst_webrtc_rtp_transceiver_direction_to_string (remote_dir));
return TRUE;
}
}
}
}
GST_LOG_OBJECT (webrtc, "no negotiation needed");
return FALSE;
}
static GstStructure *
_check_need_negotiation_task (GstWebRTCBin * webrtc, gpointer unused)
{
if (webrtc->priv->need_negotiation) {
GST_TRACE_OBJECT (webrtc, "emitting on-negotiation-needed");
PC_UNLOCK (webrtc);
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL],
0);
PC_LOCK (webrtc);
}
return NULL;
}
/* http://w3c.github.io/webrtc-pc/#dfn-update-the-negotiation-needed-flag */
static void
_update_need_negotiation (GstWebRTCBin * webrtc)
{
/* If connection's [[isClosed]] slot is true, abort these steps. */
if (webrtc->priv->is_closed)
return;
/* If connection's signaling state is not "stable", abort these steps. */
if (webrtc->signaling_state != GST_WEBRTC_SIGNALING_STATE_STABLE)
return;
/* If the result of checking if negotiation is needed is "false", clear the
* negotiation-needed flag by setting connection's [[ needNegotiation]] slot
* to false, and abort these steps. */
if (!_check_if_negotiation_is_needed (webrtc)) {
webrtc->priv->need_negotiation = FALSE;
return;
}
/* If connection's [[needNegotiation]] slot is already true, abort these steps. */
if (webrtc->priv->need_negotiation)
return;
/* Set connection's [[needNegotiation]] slot to true. */
webrtc->priv->need_negotiation = TRUE;
/* Queue a task to check connection's [[ needNegotiation]] slot and, if still
* true, fire a simple event named negotiationneeded at connection. */
gst_webrtc_bin_enqueue_task (webrtc, _check_need_negotiation_task, NULL,
NULL, NULL);
}
static GstCaps *
_query_pad_caps (GstWebRTCBin * webrtc, GstWebRTCRTPTransceiver * rtp_trans,
GstWebRTCBinPad * pad, GstCaps * filter, GError ** error)
{
GstCaps *caps;
guint i, n;
caps = gst_pad_peer_query_caps (GST_PAD (pad), filter);
GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT, caps);
/* Only return an error if actual empty caps were returned from the query. */
if (gst_caps_is_empty (caps)) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Caps negotiation on pad %s failed", GST_PAD_NAME (pad));
gst_clear_caps (&caps);
gst_caps_unref (filter);
return NULL;
}
n = gst_caps_get_size (caps);
if (n > 0) {
/* Make sure the caps are complete enough to figure out the media type and
* encoding-name, otherwise they would match with basically any media. */
caps = gst_caps_make_writable (caps);
for (i = n; i > 0; i--) {
const GstStructure *s = gst_caps_get_structure (caps, i - 1);
if (!gst_structure_has_name (s, "application/x-rtp") ||
!gst_structure_has_field (s, "media") ||
!gst_structure_has_field (s, "encoding-name")) {
gst_caps_remove_structure (caps, i - 1);
}
}
}
/* If the filtering above resulted in empty caps, or the caps were ANY to
* begin with, then don't report and error but just NULL.
*
* This would be the case if negotiation would not fail but the peer does
* not have any specific enough preferred caps that would allow us to
* use them further.
*/
if (gst_caps_is_any (caps) || gst_caps_is_empty (caps)) {
GST_DEBUG_OBJECT (webrtc, "Peer caps not specific enough");
gst_clear_caps (&caps);
}
gst_caps_unref (filter);
return caps;
}
static GstCaps *
_find_codec_preferences (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiver * rtp_trans, guint media_idx, GError ** error)
{
WebRTCTransceiver *trans = (WebRTCTransceiver *) rtp_trans;
GstCaps *ret = NULL;
GstCaps *codec_preferences = NULL;
GstWebRTCBinPad *pad = NULL;
GstPadDirection direction;
g_assert (rtp_trans);
g_assert (error && *error == NULL);
GST_LOG_OBJECT (webrtc, "retrieving codec preferences from %" GST_PTR_FORMAT,
trans);
GST_OBJECT_LOCK (rtp_trans);
if (rtp_trans->codec_preferences) {
GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT,
rtp_trans->codec_preferences);
codec_preferences = gst_caps_ref (rtp_trans->codec_preferences);
}
GST_OBJECT_UNLOCK (rtp_trans);
if (rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY)
direction = GST_PAD_SRC;
else
direction = GST_PAD_SINK;
pad = _find_pad_for_transceiver (webrtc, direction, rtp_trans);
/* try to find a pad */
if (!pad)
pad = _find_pad_for_mline (webrtc, direction, media_idx);
/* For the case where we have set our transceiver to sendrecv, but the
* sink pad has not been requested yet.
*/
if (!pad &&
rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
pad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
/* try to find a pad */
if (!pad)
pad = _find_pad_for_mline (webrtc, GST_PAD_SRC, media_idx);
}
if (pad) {
GstCaps *caps = NULL;
if (pad->received_caps) {
caps = gst_caps_ref (pad->received_caps);
} else {
static GstStaticCaps static_filter =
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) { audio, video }, payload = (int) [ 0, 127 ]");
GstCaps *filter = gst_static_caps_get (&static_filter);
filter = gst_caps_make_writable (filter);
if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "audio", NULL);
else if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "video", NULL);
caps = _query_pad_caps (webrtc, rtp_trans, pad, filter, error);
}
if (*error)
goto out;
if (caps &&
rtp_trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GstWebRTCBinPad *srcpad =
_find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
if (srcpad) {
caps = _query_pad_caps (webrtc, rtp_trans, srcpad, caps, error);
gst_object_unref (srcpad);
if (*error)
goto out;
}
}
if (caps && codec_preferences) {
GstCaps *intersection;
intersection = gst_caps_intersect_full (codec_preferences, caps,
GST_CAPS_INTERSECT_FIRST);
gst_clear_caps (&caps);
if (gst_caps_is_empty (intersection)) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Caps negotiation on pad %s failed against codec preferences",
GST_PAD_NAME (pad));
gst_clear_caps (&intersection);
} else {
caps = intersection;
}
}
if (caps) {
if (trans)
gst_caps_replace (&trans->last_retrieved_caps, caps);
ret = caps;
}
}
if (!ret) {
if (codec_preferences)
ret = gst_caps_ref (codec_preferences);
else if (trans->last_retrieved_caps)
ret = gst_caps_ref (trans->last_retrieved_caps);
}
out:
if (pad)
gst_object_unref (pad);
if (codec_preferences)
gst_caps_unref (codec_preferences);
if (!ret)
GST_DEBUG_OBJECT (trans, "Could not find caps for mline %u", media_idx);
return ret;
}
static GstCaps *
_add_supported_attributes_to_caps (GstWebRTCBin * webrtc,
WebRTCTransceiver * trans, const GstCaps * caps)
{
GstWebRTCKind kind;
GstCaps *ret;
guint i;
if (caps == NULL)
return NULL;
ret = gst_caps_make_writable (caps);
kind = webrtc_kind_from_caps (ret);
for (i = 0; i < gst_caps_get_size (ret); i++) {
GstStructure *s = gst_caps_get_structure (ret, i);
if (trans->do_nack)
if (!gst_structure_has_field (s, "rtcp-fb-nack"))
gst_structure_set (s, "rtcp-fb-nack", G_TYPE_BOOLEAN, TRUE, NULL);
if (kind == GST_WEBRTC_KIND_VIDEO) {
if (!gst_structure_has_field (s, "rtcp-fb-nack-pli"))
gst_structure_set (s, "rtcp-fb-nack-pli", G_TYPE_BOOLEAN, TRUE, NULL);
if (!gst_structure_has_field (s, "rtcp-fb-ccm-fir"))
gst_structure_set (s, "rtcp-fb-ccm-fir", G_TYPE_BOOLEAN, TRUE, NULL);
}
if (!gst_structure_has_field (s, "rtcp-fb-transport-cc"))
gst_structure_set (s, "rtcp-fb-transport-cc", G_TYPE_BOOLEAN, TRUE, NULL);
/* FIXME: codec-specific parameters? */
}
return ret;
}
static void
_on_ice_transport_notify_state (GstWebRTCICETransport * transport,
GParamSpec * pspec, GstWebRTCBin * webrtc)
{
_update_ice_connection_state (webrtc);
_update_peer_connection_state (webrtc);
}
static void
_on_local_ice_candidate_cb (GstWebRTCICE * ice, guint session_id,
gchar * candidate, GstWebRTCBin * webrtc);
static void
_on_ice_transport_notify_gathering_state (GstWebRTCICETransport * transport,
GParamSpec * pspec, GstWebRTCBin * webrtc)
{
GstWebRTCICEGatheringState ice_state;
g_object_get (transport, "gathering-state", &ice_state, NULL);
if (ice_state == GST_WEBRTC_ICE_GATHERING_STATE_COMPLETE) {
TransportStream *stream =
_find_transport_for_ice_transport (webrtc, transport);
/* signal end-of-candidates */
_on_local_ice_candidate_cb (webrtc->priv->ice, stream->session_id,
(char *) "", webrtc);
}
gst_webrtc_bin_enqueue_task (webrtc, _update_ice_gathering_state_task, NULL,
NULL, NULL);
}
static void
_on_dtls_transport_notify_state (GstWebRTCDTLSTransport * transport,
GParamSpec * pspec, GstWebRTCBin * webrtc)
{
_update_peer_connection_state (webrtc);
}
static gboolean
_on_sending_rtcp (GObject * internal_session, GstBuffer * buffer,
gboolean early, gpointer user_data)
{
GstWebRTCBin *webrtc = user_data;
GstRTCPBuffer rtcp = GST_RTCP_BUFFER_INIT;
GstRTCPPacket packet;
if (!gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp))
goto done;
if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) {
if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_SR) {
guint32 ssrc;
GstWebRTCRTPTransceiver *rtp_trans = NULL;
WebRTCTransceiver *trans;
guint rtp_session;
SsrcMapItem *mid;
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, NULL, NULL, NULL,
NULL);
rtp_session =
GPOINTER_TO_UINT (g_object_get_data (internal_session,
"GstWebRTCBinRTPSessionID"));
mid = find_mid_ssrc_for_ssrc (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, rtp_session, ssrc);
if (mid && mid->mid) {
rtp_trans = _find_transceiver_for_mid (webrtc, mid->mid);
GST_LOG_OBJECT (webrtc, "found %" GST_PTR_FORMAT " from mid entry "
"using rtp session %u ssrc %u -> mid \'%s\'", rtp_trans,
rtp_session, ssrc, mid->mid);
}
trans = (WebRTCTransceiver *) rtp_trans;
if (rtp_trans && rtp_trans->sender && trans->tos_event) {
GstPad *pad;
gchar *pad_name = NULL;
pad_name =
g_strdup_printf ("send_rtcp_src_%u",
rtp_trans->sender->transport->session_id);
pad = gst_element_get_static_pad (webrtc->rtpbin, pad_name);
g_free (pad_name);
if (pad) {
gst_pad_push_event (pad, gst_event_ref (trans->tos_event));
gst_object_unref (pad);
}
}
}
}
gst_rtcp_buffer_unmap (&rtcp);
done:
/* False means we don't care about suppression */
return FALSE;
}
static void
gst_webrtc_bin_attach_tos_to_session (GstWebRTCBin * webrtc, guint session_id)
{
GObject *internal_session = NULL;
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
session_id, &internal_session);
if (internal_session) {
g_object_set_data (internal_session, "GstWebRTCBinRTPSessionID",
GUINT_TO_POINTER (session_id));
g_signal_connect (internal_session, "on-sending-rtcp",
G_CALLBACK (_on_sending_rtcp), webrtc);
g_object_unref (internal_session);
}
}
static void
weak_free (GWeakRef * weak)
{
g_weak_ref_clear (weak);
g_free (weak);
}
static GstPadProbeReturn
_nicesink_pad_probe (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
GstWebRTCBin *webrtc = g_weak_ref_get ((GWeakRef *) user_data);
if (!webrtc)
return GST_PAD_PROBE_REMOVE;
if (GST_EVENT_TYPE (GST_PAD_PROBE_INFO_EVENT (info))
== GST_EVENT_CUSTOM_DOWNSTREAM_STICKY) {
const GstStructure *s =
gst_event_get_structure (GST_PAD_PROBE_INFO_EVENT (info));
if (gst_structure_has_name (s, "GstWebRtcBinUpdateTos")) {
const char *mid;
gint priority;
if ((mid = gst_structure_get_string (s, "mid"))) {
GstWebRTCRTPTransceiver *rtp_trans;
rtp_trans = _find_transceiver_for_mid (webrtc, mid);
if (rtp_trans) {
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
GstWebRTCICEStream *stream = _find_ice_stream_for_session (webrtc,
trans->stream->session_id);
guint8 dscp = 0;
/* Set DSCP field based on
* https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5
*/
switch (rtp_trans->sender->priority) {
case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW:
dscp = 8; /* CS1 */
break;
case GST_WEBRTC_PRIORITY_TYPE_LOW:
dscp = 0; /* DF */
break;
case GST_WEBRTC_PRIORITY_TYPE_MEDIUM:
switch (rtp_trans->kind) {
case GST_WEBRTC_KIND_AUDIO:
dscp = 46; /* EF */
break;
case GST_WEBRTC_KIND_VIDEO:
dscp = 38; /* AF43 *//* TODO: differentiate non-interactive */
break;
case GST_WEBRTC_KIND_UNKNOWN:
dscp = 0;
break;
}
break;
case GST_WEBRTC_PRIORITY_TYPE_HIGH:
switch (rtp_trans->kind) {
case GST_WEBRTC_KIND_AUDIO:
dscp = 46; /* EF */
break;
case GST_WEBRTC_KIND_VIDEO:
dscp = 36; /* AF42 *//* TODO: differentiate non-interactive */
break;
case GST_WEBRTC_KIND_UNKNOWN:
dscp = 0;
break;
}
break;
}
gst_webrtc_ice_set_tos (webrtc->priv->ice, stream, dscp << 2);
}
} else if (gst_structure_get_enum (s, "sctp-priority",
GST_TYPE_WEBRTC_PRIORITY_TYPE, &priority)) {
guint8 dscp = 0;
/* Set DSCP field based on
* https://tools.ietf.org/html/draft-ietf-tsvwg-rtcweb-qos-18#section-5
*/
switch (priority) {
case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW:
dscp = 8; /* CS1 */
break;
case GST_WEBRTC_PRIORITY_TYPE_LOW:
dscp = 0; /* DF */
break;
case GST_WEBRTC_PRIORITY_TYPE_MEDIUM:
dscp = 10; /* AF11 */
break;
case GST_WEBRTC_PRIORITY_TYPE_HIGH:
dscp = 18; /* AF21 */
break;
}
if (webrtc->priv->data_channel_transport)
gst_webrtc_ice_set_tos (webrtc->priv->ice,
webrtc->priv->data_channel_transport->stream, dscp << 2);
}
}
}
gst_object_unref (webrtc);
return GST_PAD_PROBE_OK;
}
static void gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc);
static void
gst_webrtc_bin_update_sctp_priority (GstWebRTCBin * webrtc)
{
GstWebRTCPriorityType sctp_priority = 0;
guint i;
if (!webrtc->priv->sctp_transport)
return;
DC_LOCK (webrtc);
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
GstWebRTCDataChannel *channel
= g_ptr_array_index (webrtc->priv->data_channels, i);
sctp_priority = MAX (sctp_priority, channel->priority);
}
DC_UNLOCK (webrtc);
/* Default priority is low means DSCP field is left as 0 */
if (sctp_priority == 0)
sctp_priority = GST_WEBRTC_PRIORITY_TYPE_LOW;
/* Nobody asks for DSCP, leave it as-is */
if (sctp_priority == GST_WEBRTC_PRIORITY_TYPE_LOW &&
!webrtc->priv->tos_attached)
return;
/* If one stream has a non-default priority, then everyone else does too */
gst_webrtc_bin_attach_tos (webrtc);
webrtc_sctp_transport_set_priority (webrtc->priv->sctp_transport,
sctp_priority);
}
static void
gst_webrtc_bin_attach_probe_to_ice_sink (GstWebRTCBin * webrtc,
GstWebRTCICETransport * transport)
{
GstPad *pad;
GWeakRef *weak;
pad = gst_element_get_static_pad (transport->sink, "sink");
weak = g_new0 (GWeakRef, 1);
g_weak_ref_init (weak, webrtc);
gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM,
_nicesink_pad_probe, weak, (GDestroyNotify) weak_free);
gst_object_unref (pad);
}
static void
gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc)
{
guint i;
if (webrtc->priv->tos_attached)
return;
webrtc->priv->tos_attached = TRUE;
for (i = 0; i < webrtc->priv->transports->len; i++) {
TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i);
gst_webrtc_bin_attach_tos_to_session (webrtc, stream->session_id);
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc,
stream->transport->transport);
}
gst_webrtc_bin_update_sctp_priority (webrtc);
}
static void
on_transceiver_notify_direction (GstWebRTCRTPTransceiver * transceiver,
GParamSpec * pspec, GstWebRTCBin * webrtc)
{
PC_LOCK (webrtc);
_update_need_negotiation (webrtc);
PC_UNLOCK (webrtc);
}
static WebRTCTransceiver *
_create_webrtc_transceiver (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiverDirection direction, guint mline, GstWebRTCKind kind,
GstCaps * codec_preferences)
{
WebRTCTransceiver *trans;
GstWebRTCRTPTransceiver *rtp_trans;
GstWebRTCRTPSender *sender;
GstWebRTCRTPReceiver *receiver;
sender = gst_webrtc_rtp_sender_new ();
receiver = gst_webrtc_rtp_receiver_new ();
trans = webrtc_transceiver_new (webrtc, sender, receiver);
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
rtp_trans->direction = direction;
rtp_trans->mline = mline;
rtp_trans->kind = kind;
rtp_trans->codec_preferences =
codec_preferences ? gst_caps_ref (codec_preferences) : NULL;
/* FIXME: We don't support stopping transceiver yet so they're always not stopped */
rtp_trans->stopped = FALSE;
GST_LOG_OBJECT (webrtc, "created new transceiver %" GST_PTR_FORMAT " with "
"direction %s (%d), mline %u, kind %s (%d)", rtp_trans,
gst_webrtc_rtp_transceiver_direction_to_string (direction), direction,
mline, gst_webrtc_kind_to_string (kind), kind);
g_signal_connect_object (sender, "notify::priority",
G_CALLBACK (gst_webrtc_bin_attach_tos), webrtc, G_CONNECT_SWAPPED);
g_signal_connect_object (trans, "notify::direction",
G_CALLBACK (on_transceiver_notify_direction), webrtc, G_CONNECT_DEFAULT);
g_ptr_array_add (webrtc->priv->transceivers, trans);
gst_object_unref (sender);
gst_object_unref (receiver);
return trans;
}
static TransportStream *
_create_transport_channel (GstWebRTCBin * webrtc, guint session_id)
{
GstWebRTCDTLSTransport *transport;
TransportStream *ret;
gchar *pad_name;
/* FIXME: how to parametrize the sender and the receiver */
ret = transport_stream_new (webrtc, session_id);
transport = ret->transport;
g_signal_connect (G_OBJECT (transport->transport), "notify::state",
G_CALLBACK (_on_ice_transport_notify_state), webrtc);
g_signal_connect (G_OBJECT (transport->transport),
"notify::gathering-state",
G_CALLBACK (_on_ice_transport_notify_gathering_state), webrtc);
g_signal_connect (G_OBJECT (transport), "notify::state",
G_CALLBACK (_on_dtls_transport_notify_state), webrtc);
if (webrtc->priv->tos_attached)
gst_webrtc_bin_attach_probe_to_ice_sink (webrtc, transport->transport);
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->send_bin));
gst_bin_add (GST_BIN (webrtc), GST_ELEMENT (ret->receive_bin));
g_ptr_array_add (webrtc->priv->transports, ret);
pad_name = g_strdup_printf ("recv_rtcp_sink_%u", ret->session_id);
if (!gst_element_link_pads (GST_ELEMENT (ret->receive_bin), "rtcp_src",
GST_ELEMENT (webrtc->rtpbin), pad_name))
g_warn_if_reached ();
g_free (pad_name);
pad_name = g_strdup_printf ("send_rtcp_src_%u", ret->session_id);
if (!gst_element_link_pads (GST_ELEMENT (webrtc->rtpbin), pad_name,
GST_ELEMENT (ret->send_bin), "rtcp_sink"))
g_warn_if_reached ();
g_free (pad_name);
GST_TRACE_OBJECT (webrtc,
"Create transport %" GST_PTR_FORMAT " for session %u", ret, session_id);
return ret;
}
static TransportStream *
_get_or_create_rtp_transport_channel (GstWebRTCBin * webrtc, guint session_id)
{
TransportStream *ret;
ret = _find_transport_for_session (webrtc, session_id);
if (!ret)
ret = _create_transport_channel (webrtc, session_id);
gst_element_sync_state_with_parent (GST_ELEMENT (ret->send_bin));
gst_element_sync_state_with_parent (GST_ELEMENT (ret->receive_bin));
return ret;
}
/* this is called from the webrtc thread with the pc lock held */
static void
_on_data_channel_ready_state (WebRTCDataChannel * channel,
GParamSpec * pspec, GstWebRTCBin * webrtc)
{
GstWebRTCDataChannelState ready_state;
g_object_get (channel, "ready-state", &ready_state, NULL);
if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
gboolean found;
DC_LOCK (webrtc);
found = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel);
if (found == FALSE) {
GST_FIXME_OBJECT (webrtc, "Received open for unknown data channel");
DC_UNLOCK (webrtc);
return;
}
g_ptr_array_add (webrtc->priv->data_channels, gst_object_ref (channel));
webrtc->priv->data_channels_opened++;
DC_UNLOCK (webrtc);
gst_webrtc_bin_update_sctp_priority (webrtc);
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL], 0,
channel);
} else if (ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
gboolean found_pending;
gboolean found;
DC_LOCK (webrtc);
found_pending =
g_ptr_array_remove (webrtc->priv->pending_data_channels, channel);
found = found_pending
|| g_ptr_array_remove (webrtc->priv->data_channels, channel);
if (found == FALSE) {
GST_FIXME_OBJECT (webrtc, "Received close for unknown data channel");
} else if (found_pending == FALSE) {
webrtc->priv->data_channels_closed++;
}
DC_UNLOCK (webrtc);
}
}
static void
_on_sctpdec_pad_added (GstElement * sctpdec, GstPad * pad,
GstWebRTCBin * webrtc)
{
WebRTCDataChannel *channel;
guint stream_id;
GstPad *sink_pad;
if (sscanf (GST_PAD_NAME (pad), "src_%u", &stream_id) != 1)
return;
DC_LOCK (webrtc);
channel = _find_data_channel_for_id (webrtc, stream_id);
if (!channel) {
channel = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, NULL);
channel->parent.id = stream_id;
webrtc_data_channel_set_webrtcbin (channel, webrtc);
g_signal_emit (webrtc, gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL],
0, channel, FALSE);
gst_bin_add (GST_BIN (webrtc), channel->src_bin);
gst_bin_add (GST_BIN (webrtc), channel->sink_bin);
gst_element_sync_state_with_parent (channel->src_bin);
gst_element_sync_state_with_parent (channel->sink_bin);
webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport);
g_ptr_array_add (webrtc->priv->pending_data_channels, channel);
}
DC_UNLOCK (webrtc);
g_signal_connect (channel, "notify::ready-state",
G_CALLBACK (_on_data_channel_ready_state), webrtc);
sink_pad = gst_element_get_static_pad (channel->sink_bin, "sink");
if (gst_pad_link (pad, sink_pad) != GST_PAD_LINK_OK)
GST_WARNING_OBJECT (channel, "Failed to link sctp pad %s with channel %"
GST_PTR_FORMAT, GST_PAD_NAME (pad), channel);
gst_object_unref (sink_pad);
}
static void
_on_sctp_state_notify (WebRTCSCTPTransport * sctp, GParamSpec * pspec,
GstWebRTCBin * webrtc)
{
GstWebRTCSCTPTransportState state;
g_object_get (sctp, "state", &state, NULL);
if (state == GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED) {
int i;
GST_DEBUG_OBJECT (webrtc, "SCTP association established");
DC_LOCK (webrtc);
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
WebRTCDataChannel *channel;
channel = g_ptr_array_index (webrtc->priv->data_channels, i);
webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport);
if (!channel->parent.negotiated && !channel->opened)
webrtc_data_channel_start_negotiation (channel);
}
DC_UNLOCK (webrtc);
}
}
/* Forward declaration so we can easily disconnect the signal handler */
static void _on_sctp_notify_dtls_state (GstWebRTCDTLSTransport * transport,
GParamSpec * pspec, GstWebRTCBin * webrtc);
static GstStructure *
_sctp_check_dtls_state_task (GstWebRTCBin * webrtc, gpointer unused)
{
TransportStream *stream;
GstWebRTCDTLSTransport *transport;
GstWebRTCDTLSTransportState dtls_state;
WebRTCSCTPTransport *sctp_transport;
stream = webrtc->priv->data_channel_transport;
transport = stream->transport;
g_object_get (transport, "state", &dtls_state, NULL);
/* Not connected yet so just return */
if (dtls_state != GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED) {
GST_DEBUG_OBJECT (webrtc,
"Data channel DTLS connection is not ready yet: %d", dtls_state);
return NULL;
}
GST_DEBUG_OBJECT (webrtc, "Data channel DTLS connection is now ready");
sctp_transport = webrtc->priv->sctp_transport;
/* Not locked state anymore so this was already taken care of before */
if (!gst_element_is_locked_state (sctp_transport->sctpdec))
return NULL;
/* Start up the SCTP elements now that the DTLS connection is established */
gst_element_set_locked_state (sctp_transport->sctpdec, FALSE);
gst_element_set_locked_state (sctp_transport->sctpenc, FALSE);
gst_element_sync_state_with_parent (GST_ELEMENT (sctp_transport->sctpdec));
gst_element_sync_state_with_parent (GST_ELEMENT (sctp_transport->sctpenc));
if (sctp_transport->sctpdec_block_id) {
GstPad *receive_srcpad;
receive_srcpad =
gst_element_get_static_pad (GST_ELEMENT (stream->receive_bin),
"data_src");
gst_pad_remove_probe (receive_srcpad, sctp_transport->sctpdec_block_id);
sctp_transport->sctpdec_block_id = 0;
gst_object_unref (receive_srcpad);
}
g_signal_handlers_disconnect_by_func (transport, _on_sctp_notify_dtls_state,
webrtc);
return NULL;
}
static void
_on_sctp_notify_dtls_state (GstWebRTCDTLSTransport * transport,
GParamSpec * pspec, GstWebRTCBin * webrtc)
{
GstWebRTCDTLSTransportState dtls_state;
g_object_get (transport, "state", &dtls_state, NULL);
GST_TRACE_OBJECT (webrtc, "Data channel DTLS state changed to %d",
dtls_state);
/* Connected now, so schedule a task to update the state of the SCTP
* elements */
if (dtls_state == GST_WEBRTC_DTLS_TRANSPORT_STATE_CONNECTED) {
gst_webrtc_bin_enqueue_task (webrtc,
(GstWebRTCBinFunc) _sctp_check_dtls_state_task, NULL, NULL, NULL);
}
}
static GstPadProbeReturn
sctp_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;
}
static TransportStream *
_get_or_create_data_channel_transports (GstWebRTCBin * webrtc, guint session_id)
{
if (!webrtc->priv->data_channel_transport) {
TransportStream *stream;
WebRTCSCTPTransport *sctp_transport;
stream = _find_transport_for_session (webrtc, session_id);
if (!stream)
stream = _create_transport_channel (webrtc, session_id);
webrtc->priv->data_channel_transport = stream;
if (!(sctp_transport = webrtc->priv->sctp_transport)) {
sctp_transport = webrtc_sctp_transport_new ();
sctp_transport->transport =
g_object_ref (webrtc->priv->data_channel_transport->transport);
sctp_transport->webrtcbin = webrtc;
/* Don't automatically start SCTP elements as part of webrtcbin. We
* need to delay this until the DTLS transport is fully connected! */
gst_element_set_locked_state (sctp_transport->sctpdec, TRUE);
gst_element_set_locked_state (sctp_transport->sctpenc, TRUE);
gst_bin_add (GST_BIN (webrtc), sctp_transport->sctpdec);
gst_bin_add (GST_BIN (webrtc), sctp_transport->sctpenc);
}
g_signal_connect (sctp_transport->sctpdec, "pad-added",
G_CALLBACK (_on_sctpdec_pad_added), webrtc);
g_signal_connect (sctp_transport, "notify::state",
G_CALLBACK (_on_sctp_state_notify), webrtc);
if (sctp_transport->sctpdec_block_id == 0) {
GstPad *receive_srcpad;
receive_srcpad =
gst_element_get_static_pad (GST_ELEMENT (stream->receive_bin),
"data_src");
sctp_transport->sctpdec_block_id =
gst_pad_add_probe (receive_srcpad,
GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_DATA_DOWNSTREAM,
(GstPadProbeCallback) sctp_pad_block, NULL, NULL);
gst_object_unref (receive_srcpad);
}
if (!gst_element_link_pads (GST_ELEMENT (stream->receive_bin), "data_src",
GST_ELEMENT (sctp_transport->sctpdec), "sink"))
g_warn_if_reached ();
if (!gst_element_link_pads (GST_ELEMENT (sctp_transport->sctpenc), "src",
GST_ELEMENT (stream->send_bin), "data_sink"))
g_warn_if_reached ();
gst_element_sync_state_with_parent (GST_ELEMENT (stream->send_bin));
gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
if (!webrtc->priv->sctp_transport) {
/* Connect to the notify::state signal to get notified when the DTLS
* connection is established. Only then can we start the SCTP elements */
g_signal_connect (stream->transport, "notify::state",
G_CALLBACK (_on_sctp_notify_dtls_state), webrtc);
/* As this would be racy otherwise, also schedule a task that checks the
* current state of the connection already without getting the signal
* called */
gst_webrtc_bin_enqueue_task (webrtc,
(GstWebRTCBinFunc) _sctp_check_dtls_state_task, NULL, NULL, NULL);
}
webrtc->priv->sctp_transport = sctp_transport;
gst_webrtc_bin_update_sctp_priority (webrtc);
}
return webrtc->priv->data_channel_transport;
}
static TransportStream *
_get_or_create_transport_stream (GstWebRTCBin * webrtc, guint session_id,
gboolean is_datachannel)
{
if (is_datachannel)
return _get_or_create_data_channel_transports (webrtc, session_id);
else
return _get_or_create_rtp_transport_channel (webrtc, session_id);
}
struct media_payload_map_item
{
guint media_pt;
guint red_pt;
guint ulpfec_pt;
guint rtx_pt;
guint red_rtx_pt;
};
static void
media_payload_map_item_init (struct media_payload_map_item *item,
guint media_pt)
{
item->media_pt = media_pt;
item->red_pt = G_MAXUINT;
item->rtx_pt = G_MAXUINT;
item->ulpfec_pt = G_MAXUINT;
item->red_rtx_pt = G_MAXUINT;
}
static struct media_payload_map_item *
find_payload_map_for_media_pt (GArray * media_mapping, guint media_pt)
{
guint i;
for (i = 0; i < media_mapping->len; i++) {
struct media_payload_map_item *item;
item = &g_array_index (media_mapping, struct media_payload_map_item, i);
if (item->media_pt == media_pt)
return item;
}
return NULL;
}
static struct media_payload_map_item *
find_or_create_payload_map_for_media_pt (GArray * media_mapping, guint media_pt)
{
struct media_payload_map_item new_item;
struct media_payload_map_item *item;
if ((item = find_payload_map_for_media_pt (media_mapping, media_pt)))
return item;
media_payload_map_item_init (&new_item, media_pt);
g_array_append_val (media_mapping, new_item);
return &g_array_index (media_mapping, struct media_payload_map_item,
media_mapping->len - 1);
}
static gboolean
_pick_available_pt (GArray * media_mapping, guint * ret)
{
int i;
for (i = 96; i <= 127; i++) {
gboolean available = TRUE;
int j;
for (j = 0; j < media_mapping->len; j++) {
struct media_payload_map_item *item;
item = &g_array_index (media_mapping, struct media_payload_map_item, j);
if (item->media_pt == i || item->red_pt == i || item->rtx_pt == i
|| item->ulpfec_pt == i || item->red_rtx_pt == i) {
available = FALSE;
break;
}
}
if (available) {
*ret = i;
return TRUE;
}
}
*ret = G_MAXUINT;
return FALSE;
}
static gboolean
_pick_fec_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans,
GArray * media_mapping, gint clockrate, gint media_pt, gint * rtx_target_pt,
GstSDPMedia * media)
{
gboolean ret = TRUE;
if (trans->fec_type == GST_WEBRTC_FEC_TYPE_NONE)
goto done;
if (trans->fec_type == GST_WEBRTC_FEC_TYPE_ULP_RED && clockrate != -1) {
struct media_payload_map_item *item;
gchar *str;
item = find_or_create_payload_map_for_media_pt (media_mapping, media_pt);
if (item->red_pt == G_MAXUINT) {
if (!(ret = _pick_available_pt (media_mapping, &item->red_pt)))
goto done;
}
/* https://tools.ietf.org/html/rfc5109#section-14.1 */
str = g_strdup_printf ("%u", item->red_pt);
gst_sdp_media_add_format (media, str);
g_free (str);
str = g_strdup_printf ("%u red/%d", item->red_pt, clockrate);
gst_sdp_media_add_attribute (media, "rtpmap", str);
g_free (str);
*rtx_target_pt = item->red_pt;
if (item->ulpfec_pt == G_MAXUINT) {
if (!(ret = _pick_available_pt (media_mapping, &item->ulpfec_pt)))
goto done;
}
str = g_strdup_printf ("%u", item->ulpfec_pt);
gst_sdp_media_add_format (media, str);
g_free (str);
str = g_strdup_printf ("%u ulpfec/%d", item->ulpfec_pt, clockrate);
gst_sdp_media_add_attribute (media, "rtpmap", str);
g_free (str);
}
done:
return ret;
}
static void
add_rtx_to_media (WebRTCTransceiver * trans, gint clockrate, gint rtx_pt,
gint rtx_target_pt, guint target_ssrc, GstSDPMedia * media)
{
char *str;
/* https://tools.ietf.org/html/rfc4588#section-8.6 */
if (target_ssrc != -1) {
str = g_strdup_printf ("%u", target_ssrc);
gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT,
g_random_int (), NULL);
g_free (str);
}
str = g_strdup_printf ("%u", rtx_pt);
gst_sdp_media_add_format (media, str);
g_free (str);
str = g_strdup_printf ("%u rtx/%d", rtx_pt, clockrate);
gst_sdp_media_add_attribute (media, "rtpmap", str);
g_free (str);
str = g_strdup_printf ("%u apt=%d", rtx_pt, rtx_target_pt);
gst_sdp_media_add_attribute (media, "fmtp", str);
g_free (str);
}
static gboolean
_pick_rtx_payload_types (GstWebRTCBin * webrtc, WebRTCTransceiver * trans,
GArray * media_mapping, gint clockrate, gint media_pt, gint target_pt,
guint target_ssrc, GstSDPMedia * media)
{
gboolean ret = TRUE;
if (trans->local_rtx_ssrc_map)
gst_structure_free (trans->local_rtx_ssrc_map);
trans->local_rtx_ssrc_map =
gst_structure_new_empty ("application/x-rtp-ssrc-map");
if (trans->do_nack) {
struct media_payload_map_item *item;
item = find_or_create_payload_map_for_media_pt (media_mapping, media_pt);
if (item->rtx_pt == G_MAXUINT) {
if (!(ret = _pick_available_pt (media_mapping, &item->rtx_pt)))
goto done;
}
add_rtx_to_media (trans, clockrate, item->rtx_pt, media_pt, target_ssrc,
media);
if (item->red_pt != G_MAXUINT) {
/* Workaround chrome bug: https://bugs.chromium.org/p/webrtc/issues/detail?id=6196 */
if (item->red_rtx_pt == G_MAXUINT) {
if (!(ret = _pick_available_pt (media_mapping, &item->red_rtx_pt)))
goto done;
}
add_rtx_to_media (trans, clockrate, item->red_rtx_pt, item->red_pt,
target_ssrc, media);
}
}
done:
return ret;
}
/* https://tools.ietf.org/html/rfc5576#section-4.2 */
static gboolean
_media_add_rtx_ssrc_group (GQuark field_id, const GValue * value,
GstSDPMedia * media)
{
gchar *str;
str =
g_strdup_printf ("FID %s %u", g_quark_to_string (field_id),
g_value_get_uint (value));
gst_sdp_media_add_attribute (media, "ssrc-group", str);
g_free (str);
return TRUE;
}
typedef struct
{
GstSDPMedia *media;
GstWebRTCBin *webrtc;
WebRTCTransceiver *trans;
} RtxSsrcData;
static gboolean
_media_add_rtx_ssrc (GQuark field_id, const GValue * value, RtxSsrcData * data)
{
gchar *str;
GstStructure *sdes;
const gchar *cname;
GstWebRTCBinPad *sink_pad;
const char *msid = NULL;
g_object_get (data->webrtc->rtpbin, "sdes", &sdes, NULL);
/* http://www.freesoft.org/CIE/RFC/1889/24.htm */
cname = gst_structure_get_string (sdes, "cname");
sink_pad =
_find_pad_for_transceiver (data->webrtc, GST_PAD_SINK,
GST_WEBRTC_RTP_TRANSCEIVER (data->trans));
if (sink_pad)
msid = sink_pad->msid;
/* fallback to cname if no msid provided */
if (!msid)
msid = cname;
/* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */
/* FIXME: the ssrc is not present in RFC8830, do we still need that? */
str =
g_strdup_printf ("%u msid:%s %s", g_value_get_uint (value),
msid, GST_OBJECT_NAME (data->trans));
gst_sdp_media_add_attribute (data->media, "ssrc", str);
g_free (str);
str = g_strdup_printf ("%u cname:%s", g_value_get_uint (value), cname);
gst_sdp_media_add_attribute (data->media, "ssrc", str);
g_free (str);
gst_clear_object (&sink_pad);
gst_structure_free (sdes);
return TRUE;
}
static void
_media_add_ssrcs (GstSDPMedia * media, GstCaps * caps, GstWebRTCBin * webrtc,
WebRTCTransceiver * trans)
{
guint i;
RtxSsrcData data = { media, webrtc, trans };
const gchar *cname;
GstStructure *sdes;
g_object_get (webrtc->rtpbin, "sdes", &sdes, NULL);
/* http://www.freesoft.org/CIE/RFC/1889/24.htm */
cname = gst_structure_get_string (sdes, "cname");
if (trans->local_rtx_ssrc_map)
gst_structure_foreach (trans->local_rtx_ssrc_map,
(GstStructureForeachFunc) _media_add_rtx_ssrc_group, media);
for (i = 0; i < gst_caps_get_size (caps); i++) {
const GstStructure *s = gst_caps_get_structure (caps, i);
guint ssrc;
if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
gchar *str;
GstWebRTCBinPad *sink_pad;
const char *msid = NULL;
sink_pad =
_find_pad_for_transceiver (webrtc, GST_PAD_SINK,
GST_WEBRTC_RTP_TRANSCEIVER (trans));
if (sink_pad)
msid = sink_pad->msid;
/* fallback to cname if no msid provided */
if (!msid)
msid = cname;
/* https://tools.ietf.org/html/draft-ietf-mmusic-msid-16 */
/* FIXME: the ssrc is not present in RFC8830, do we still need that? */
str =
g_strdup_printf ("%u msid:%s %s", ssrc, msid,
GST_OBJECT_NAME (trans));
gst_sdp_media_add_attribute (media, "ssrc", str);
g_free (str);
str = g_strdup_printf ("%u cname:%s", ssrc, cname);
gst_sdp_media_add_attribute (media, "ssrc", str);
g_free (str);
gst_clear_object (&sink_pad);
}
}
gst_structure_free (sdes);
if (trans->local_rtx_ssrc_map)
gst_structure_foreach (trans->local_rtx_ssrc_map,
(GstStructureForeachFunc) _media_add_rtx_ssrc, &data);
}
static void
_add_fingerprint_to_media (GstWebRTCDTLSTransport * transport,
GstSDPMedia * media)
{
gchar *cert, *fingerprint, *val;
g_object_get (transport, "certificate", &cert, NULL);
fingerprint =
_generate_fingerprint_from_certificate (cert, G_CHECKSUM_SHA256);
g_free (cert);
val =
g_strdup_printf ("%s %s",
_g_checksum_to_webrtc_string (G_CHECKSUM_SHA256), fingerprint);
g_free (fingerprint);
gst_sdp_media_add_attribute (media, "fingerprint", val);
g_free (val);
}
static gchar *
_parse_extmap (GQuark field_id, const GValue * value, GError ** error)
{
gchar *ret = NULL;
if (G_VALUE_HOLDS_STRING (value)) {
ret = g_value_dup_string (value);
} else if (G_VALUE_HOLDS (value, GST_TYPE_ARRAY)
&& gst_value_array_get_size (value) == 3) {
const GValue *val;
const gchar *direction, *extensionname, *extensionattributes;
val = gst_value_array_get_value (value, 0);
direction = g_value_get_string (val);
val = gst_value_array_get_value (value, 1);
extensionname = g_value_get_string (val);
val = gst_value_array_get_value (value, 2);
extensionattributes = g_value_get_string (val);
if (!extensionname || *extensionname == '\0')
goto done;
if (direction && *direction != '\0' && extensionattributes
&& *extensionattributes != '\0') {
ret =
g_strdup_printf ("/%s %s %s", direction, extensionname,
extensionattributes);
} else if (direction && *direction != '\0') {
ret = g_strdup_printf ("/%s %s", direction, extensionname);
} else if (extensionattributes && *extensionattributes != '\0') {
ret = g_strdup_printf ("%s %s", extensionname, extensionattributes);
} else {
ret = g_strdup (extensionname);
}
}
if (!ret && error) {
gchar *val_str = gst_value_serialize (value);
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Invalid value for %s: %s", g_quark_to_string (field_id), val_str);
g_free (val_str);
}
done:
return ret;
}
typedef struct
{
gboolean ret;
GstStructure *extmap;
GError **error;
} ExtmapData;
static gboolean
_dedup_extmap_field (GQuark field_id, const GValue * value, ExtmapData * data)
{
gboolean is_extmap =
g_str_has_prefix (g_quark_to_string (field_id), "extmap-");
if (!data->ret)
goto done;
if (is_extmap) {
gchar *new_value = _parse_extmap (field_id, value, data->error);
if (!new_value) {
data->ret = FALSE;
goto done;
}
if (gst_structure_id_has_field (data->extmap, field_id)) {
gchar *old_value =
_parse_extmap (field_id, gst_structure_id_get_value (data->extmap,
field_id), NULL);
g_assert (old_value);
if (g_strcmp0 (new_value, old_value)) {
GST_ERROR
("extmap contains different values for id %s (%s != %s)",
g_quark_to_string (field_id), old_value, new_value);
g_set_error (data->error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"extmap contains different values for id %s (%s != %s)",
g_quark_to_string (field_id), old_value, new_value);
data->ret = FALSE;
}
g_free (old_value);
}
if (data->ret) {
gst_structure_id_set_value (data->extmap, field_id, value);
}
g_free (new_value);
}
done:
return !is_extmap;
}
static GstStructure *
_gather_extmap (GstCaps * caps, GError ** error)
{
ExtmapData edata =
{ TRUE, gst_structure_new_empty ("application/x-extmap"), error };
guint i, n;
n = gst_caps_get_size (caps);
for (i = 0; i < n; i++) {
GstStructure *s = gst_caps_get_structure (caps, i);
gst_structure_filter_and_map_in_place (s,
(GstStructureFilterMapFunc) _dedup_extmap_field, &edata);
if (!edata.ret) {
gst_clear_structure (&edata.extmap);
break;
}
}
return edata.extmap;
}
struct hdrext_id
{
const char *rtphdrext_uri;
guint ext_id;
};
static gboolean
structure_value_get_rtphdrext_id (GQuark field_id, const GValue * value,
gpointer user_data)
{
struct hdrext_id *rtphdrext = user_data;
const char *field_name = g_quark_to_string (field_id);
if (g_str_has_prefix (field_name, "extmap-")) {
const char *val = NULL;
if (GST_VALUE_HOLDS_ARRAY (value) && gst_value_array_get_size (value) >= 2) {
value = gst_value_array_get_value (value, 1);
}
if (G_VALUE_HOLDS_STRING (value)) {
val = g_value_get_string (value);
}
if (g_strcmp0 (val, rtphdrext->rtphdrext_uri) == 0) {
gint64 id = g_ascii_strtoll (&field_name[strlen ("extmap-")], NULL, 10);
if (id > 0 && id < 256)
rtphdrext->ext_id = id;
return FALSE;
}
}
return TRUE;
}
// Returns -1 when not found
static guint
caps_get_rtp_header_extension_id (const GstCaps * caps,
const char *rtphdrext_uri)
{
guint i, n;
n = gst_caps_get_size (caps);
for (i = 0; i < n; i++) {
const GstStructure *s = gst_caps_get_structure (caps, i);
struct hdrext_id data = { rtphdrext_uri, -1 };
gst_structure_foreach (s, structure_value_get_rtphdrext_id, &data);
if (data.ext_id != -1)
return data.ext_id;
}
return -1;
}
static gboolean
caps_contain_rtp_header_extension (const GstCaps * caps,
const char *rtphdrext_uri)
{
return caps_get_rtp_header_extension_id (caps, rtphdrext_uri) != -1;
}
static gboolean
_copy_field (GQuark field_id, const GValue * value, GstStructure * s)
{
gst_structure_id_set_value (s, field_id, value);
return TRUE;
}
/* based off https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-18#section-5.2.1 */
static gboolean
sdp_media_from_transceiver (GstWebRTCBin * webrtc, GstSDPMedia * media,
const GstSDPMedia * last_media, GstWebRTCRTPTransceiver * trans,
guint media_idx, GString * bundled_mids, guint bundle_idx,
gchar * bundle_ufrag, gchar * bundle_pwd, GArray * media_mapping,
GHashTable * all_mids, gboolean * no_more_mlines, GError ** error)
{
/* TODO:
* rtp header extensions
* ice attributes
* rtx
* fec
* msid-semantics
* msid
* dtls fingerprints
* multiple dtls fingerprints https://tools.ietf.org/html/draft-ietf-mmusic-4572-update-05
*/
GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
gchar *ufrag, *pwd, *mid = NULL;
gboolean bundle_only;
guint rtp_session_idx;
GstCaps *caps;
GstStructure *extmap;
int i;
if (trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE)
return FALSE;
g_assert (trans->mline == -1 || trans->mline == media_idx);
rtp_session_idx = bundled_mids ? bundle_idx : media_idx;
bundle_only = bundled_mids && bundle_idx != media_idx
&& webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE;
caps = _find_codec_preferences (webrtc, trans, media_idx, error);
caps = _add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans),
caps);
if (!caps || gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
gst_clear_caps (&caps);
if (last_media) {
guint i, n;
n = gst_sdp_media_formats_len (last_media);
if (n > 0) {
caps = gst_caps_new_empty ();
for (i = 0; i < n; i++) {
guint fmt = atoi (gst_sdp_media_get_format (last_media, i));
GstCaps *tmp = gst_sdp_media_get_caps_from_media (last_media, fmt);
GstStructure *s = gst_caps_get_structure (tmp, 0);
gst_structure_set_name (s, "application/x-rtp");
gst_caps_append_structure (caps, gst_structure_copy (s));
gst_clear_caps (&tmp);
}
GST_DEBUG_OBJECT (webrtc, "using previously negotiated caps for "
"transceiver %" GST_PTR_FORMAT " %" GST_PTR_FORMAT, trans, caps);
}
}
if (!caps) {
if (WEBRTC_TRANSCEIVER (trans)->mline_locked) {
GST_WARNING_OBJECT (webrtc,
"Transceiver <%s> with mid %s has locked mline %u, but no caps. "
"Can't add more lines after this one.", GST_OBJECT_NAME (trans),
trans->mid, trans->mline);
*no_more_mlines = TRUE;
} else {
GST_WARNING_OBJECT (webrtc, "no caps available for transceiver %"
GST_PTR_FORMAT ", skipping", trans);
}
return FALSE;
}
}
if (last_media) {
const char *setup = gst_sdp_media_get_attribute_val (last_media, "setup");
if (setup) {
gst_sdp_media_add_attribute (media, "setup", setup);
} else {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
"media %u cannot renegotiate without an existing a=setup line",
media_idx);
return FALSE;
}
} else {
/* mandated by JSEP */
gst_sdp_media_add_attribute (media, "setup", "actpass");
}
/* FIXME: deal with ICE restarts */
if (last_offer && trans->mline != -1 && trans->mid) {
ufrag = g_strdup (_media_get_ice_ufrag (last_offer, trans->mline));
pwd = g_strdup (_media_get_ice_pwd (last_offer, trans->mline));
GST_DEBUG_OBJECT (trans, "%u Using previous ice parameters", media_idx);
} else {
GST_DEBUG_OBJECT (trans,
"%u Generating new ice parameters mline %i, mid %s", media_idx,
trans->mline, trans->mid);
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
_generate_ice_credentials (&ufrag, &pwd);
} else {
g_assert (bundle_ufrag && bundle_pwd);
ufrag = g_strdup (bundle_ufrag);
pwd = g_strdup (bundle_pwd);
}
}
gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag);
gst_sdp_media_add_attribute (media, "ice-pwd", pwd);
g_free (ufrag);
g_free (pwd);
gst_sdp_media_set_port_info (media, bundle_only || trans->stopped ? 0 : 9, 0);
gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0);
if (bundle_only) {
gst_sdp_media_add_attribute (media, "bundle-only", NULL);
}
/* FIXME: negotiate this */
/* FIXME: when bundle_only, these should not be added:
* https://tools.ietf.org/html/draft-ietf-mmusic-sdp-bundle-negotiation-52#section-7.1.3
* However, this causes incompatibilities with current versions
* of the major browsers */
gst_sdp_media_add_attribute (media, "rtcp-mux", "");
gst_sdp_media_add_attribute (media, "rtcp-rsize", NULL);
gst_sdp_media_add_attribute (media,
gst_webrtc_rtp_transceiver_direction_to_string (trans->direction), "");
caps = gst_caps_make_writable (caps);
/* When an extmap is defined twice for the same ID, firefox complains and
* errors out (chrome is smart enough to accept strict duplicates).
*
* To work around this, we deduplicate extmap attributes, and also error
* out when a different extmap is defined for the same ID.
*
* _gather_extmap will strip out all extmap- fields, which will then be
* added upon adding the first format for the media.
*/
extmap = _gather_extmap (caps, error);
if (!extmap) {
GST_ERROR_OBJECT (webrtc,
"Failed to build extmap for transceiver %" GST_PTR_FORMAT, trans);
gst_caps_unref (caps);
return FALSE;
}
caps = _add_supported_attributes_to_caps (webrtc, WEBRTC_TRANSCEIVER (trans),
caps);
for (i = 0; i < gst_caps_get_size (caps); i++) {
GstCaps *format = gst_caps_new_empty ();
GstStructure *s = gst_structure_copy (gst_caps_get_structure (caps, i));
if (i == 0) {
gst_structure_foreach (extmap, (GstStructureForeachFunc) _copy_field, s);
}
gst_caps_append_structure (format, s);
GST_DEBUG_OBJECT (webrtc, "Adding %u-th caps %" GST_PTR_FORMAT
" to %u-th media", i, format, media_idx);
/* this only looks at the first structure so we loop over the given caps
* and add each structure inside it piecemeal */
if (gst_sdp_media_set_media_from_caps (format, media) != GST_SDP_OK) {
GST_ERROR_OBJECT (webrtc,
"Failed to build media from caps %" GST_PTR_FORMAT
" for transceiver %" GST_PTR_FORMAT, format, trans);
gst_caps_unref (caps);
gst_caps_unref (format);
gst_structure_free (extmap);
return FALSE;
}
gst_caps_unref (format);
}
gst_clear_structure (&extmap);
{
const GstStructure *s = gst_caps_get_structure (caps, 0);
gint clockrate = -1;
gint rtx_target_pt;
guint rtx_target_ssrc = -1;
gint media_pt;
if (gst_structure_get_int (s, "payload", &media_pt) &&
webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE)
find_or_create_payload_map_for_media_pt (media_mapping, media_pt);
rtx_target_pt = media_pt;
if (!gst_structure_get_int (s, "clock-rate", &clockrate))
GST_WARNING_OBJECT (webrtc,
"Caps %" GST_PTR_FORMAT " are missing clock-rate", caps);
if (!gst_structure_get_uint (s, "ssrc", &rtx_target_ssrc)) {
if (!caps_contain_rtp_header_extension (caps, RTPHDREXT_MID)) {
GST_WARNING_OBJECT (webrtc, "Caps %" GST_PTR_FORMAT " are missing ssrc",
caps);
}
}
_pick_fec_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), media_mapping,
clockrate, media_pt, &rtx_target_pt, media);
_pick_rtx_payload_types (webrtc, WEBRTC_TRANSCEIVER (trans), media_mapping,
clockrate, media_pt, rtx_target_pt, rtx_target_ssrc, media);
}
_media_add_ssrcs (media, caps, webrtc, WEBRTC_TRANSCEIVER (trans));
/* Some identifier; we also add the media name to it so it's identifiable */
if (trans->mid) {
const char *media_mid = gst_sdp_media_get_attribute_val (media, "mid");
if (!media_mid) {
gst_sdp_media_add_attribute (media, "mid", trans->mid);
} else if (g_strcmp0 (media_mid, trans->mid) != 0) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INVALID_MODIFICATION,
"Cannot change media %u mid value from \'%s\' to \'%s\'",
media_idx, media_mid, trans->mid);
return FALSE;
}
mid = g_strdup (trans->mid);
g_hash_table_insert (all_mids, g_strdup (mid), NULL);
}
if (mid == NULL) {
const GstStructure *s = gst_caps_get_structure (caps, 0);
mid = g_strdup (gst_structure_get_string (s, "a-mid"));
if (mid) {
if (g_hash_table_contains (all_mids, (gpointer) mid)) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Cannot re-use mid \'%s\' from the caps in m= line %u that has "
"already been used for a previous m= line in the SDP", mid,
media_idx);
return FALSE;
}
g_free (WEBRTC_TRANSCEIVER (trans)->pending_mid);
WEBRTC_TRANSCEIVER (trans)->pending_mid = g_strdup (mid);
g_hash_table_insert (all_mids, g_strdup (mid), NULL);
}
}
if (mid == NULL) {
mid = g_strdup (WEBRTC_TRANSCEIVER (trans)->pending_mid);
if (mid) {
/* If it's already used, just ignore the pending one and generate
* a new one */
if (g_hash_table_contains (all_mids, (gpointer) mid)) {
g_clear_pointer (&mid, free);
g_clear_pointer (&WEBRTC_TRANSCEIVER (trans)->pending_mid, free);
} else {
gst_sdp_media_add_attribute (media, "mid", mid);
g_hash_table_insert (all_mids, g_strdup (mid), NULL);
}
}
}
if (mid == NULL) {
/* Make sure to avoid mid collisions */
while (TRUE) {
mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
webrtc->priv->media_counter++);
if (g_hash_table_contains (all_mids, (gpointer) mid)) {
g_free (mid);
} else {
gst_sdp_media_add_attribute (media, "mid", mid);
g_hash_table_insert (all_mids, g_strdup (mid), NULL);
WEBRTC_TRANSCEIVER (trans)->pending_mid = g_strdup (mid);
break;
}
}
}
/* TODO:
* - add a=candidate lines for gathered candidates
*/
if (trans->sender) {
if (!trans->sender->transport) {
TransportStream *item;
item = _get_or_create_transport_stream (webrtc, rtp_session_idx, FALSE);
webrtc_transceiver_set_transport (WEBRTC_TRANSCEIVER (trans), item);
}
_add_fingerprint_to_media (trans->sender->transport, media);
}
if (bundled_mids) {
g_assert (mid);
g_string_append_printf (bundled_mids, " %s", mid);
}
g_clear_pointer (&mid, g_free);
gst_caps_unref (caps);
return TRUE;
}
static void
gather_pad_pt (GstWebRTCBinPad * pad, GArray * media_mapping)
{
if (pad->received_caps) {
GstStructure *s = gst_caps_get_structure (pad->received_caps, 0);
gint pt;
if (gst_structure_get_int (s, "payload", &pt)) {
GST_TRACE_OBJECT (pad, "have media pt %u from received caps", pt);
find_or_create_payload_map_for_media_pt (media_mapping, pt);
}
}
}
static GArray *
gather_media_mapping (GstWebRTCBin * webrtc)
{
GstElement *element = GST_ELEMENT (webrtc);
GArray *media_mapping =
g_array_new (FALSE, FALSE, sizeof (struct media_payload_map_item));
guint i;
GST_OBJECT_LOCK (webrtc);
g_list_foreach (element->sinkpads, (GFunc) gather_pad_pt, media_mapping);
g_list_foreach (webrtc->priv->pending_pads, (GFunc) gather_pad_pt,
media_mapping);
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *trans;
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
GST_OBJECT_LOCK (trans);
if (trans->codec_preferences) {
guint j, n;
gint pt;
n = gst_caps_get_size (trans->codec_preferences);
for (j = 0; j < n; j++) {
GstStructure *s = gst_caps_get_structure (trans->codec_preferences, j);
if (gst_structure_get_int (s, "payload", &pt)) {
GST_TRACE_OBJECT (trans, "have media pt %u from codec preferences",
pt);
find_or_create_payload_map_for_media_pt (media_mapping, pt);
}
}
}
GST_OBJECT_UNLOCK (trans);
}
GST_OBJECT_UNLOCK (webrtc);
return media_mapping;
}
static gboolean
_add_data_channel_offer (GstWebRTCBin * webrtc, GstSDPMessage * msg,
GstSDPMedia * media, GString * bundled_mids, guint bundle_idx,
gchar * bundle_ufrag, gchar * bundle_pwd, GHashTable * all_mids)
{
GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
gchar *ufrag, *pwd, *sdp_mid;
gboolean bundle_only = bundled_mids
&& webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE
&& gst_sdp_message_medias_len (msg) != bundle_idx;
guint last_data_index = G_MAXUINT;
/* add data channel support */
if (webrtc->priv->data_channels->len == 0)
return FALSE;
if (last_offer) {
last_data_index = _message_get_datachannel_index (last_offer);
if (last_data_index < G_MAXUINT) {
g_assert (last_data_index < gst_sdp_message_medias_len (last_offer));
/* XXX: is this always true when recycling transceivers?
* i.e. do we always put the data channel in the same mline */
g_assert (last_data_index == gst_sdp_message_medias_len (msg));
}
}
/* mandated by JSEP */
gst_sdp_media_add_attribute (media, "setup", "actpass");
/* FIXME: only needed when restarting ICE */
if (last_offer && last_data_index < G_MAXUINT) {
ufrag = g_strdup (_media_get_ice_ufrag (last_offer, last_data_index));
pwd = g_strdup (_media_get_ice_pwd (last_offer, last_data_index));
} else {
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
_generate_ice_credentials (&ufrag, &pwd);
} else {
ufrag = g_strdup (bundle_ufrag);
pwd = g_strdup (bundle_pwd);
}
}
gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag);
gst_sdp_media_add_attribute (media, "ice-pwd", pwd);
g_free (ufrag);
g_free (pwd);
gst_sdp_media_set_media (media, "application");
gst_sdp_media_set_port_info (media, bundle_only ? 0 : 9, 0);
gst_sdp_media_set_proto (media, "UDP/DTLS/SCTP");
gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0);
gst_sdp_media_add_format (media, "webrtc-datachannel");
if (bundle_idx != gst_sdp_message_medias_len (msg))
gst_sdp_media_add_attribute (media, "bundle-only", NULL);
if (last_offer && last_data_index < G_MAXUINT) {
const GstSDPMedia *last_data_media;
const gchar *mid;
last_data_media = gst_sdp_message_get_media (last_offer, last_data_index);
mid = gst_sdp_media_get_attribute_val (last_data_media, "mid");
gst_sdp_media_add_attribute (media, "mid", mid);
} else {
/* Make sure to avoid mid collisions */
while (TRUE) {
sdp_mid = g_strdup_printf ("%s%u", gst_sdp_media_get_media (media),
webrtc->priv->media_counter++);
if (g_hash_table_contains (all_mids, (gpointer) sdp_mid)) {
g_free (sdp_mid);
} else {
gst_sdp_media_add_attribute (media, "mid", sdp_mid);
g_hash_table_insert (all_mids, sdp_mid, NULL);
break;
}
}
}
if (bundled_mids) {
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
g_assert (mid);
g_string_append_printf (bundled_mids, " %s", mid);
}
/* FIXME: negotiate this properly */
gst_sdp_media_add_attribute (media, "sctp-port", "5000");
_get_or_create_data_channel_transports (webrtc,
bundled_mids ? 0 : webrtc->priv->transceivers->len);
_add_fingerprint_to_media (webrtc->priv->sctp_transport->transport, media);
return TRUE;
}
/* TODO: use the options argument */
static GstSDPMessage *
_create_offer_task (GstWebRTCBin * webrtc, const GstStructure * options,
GError ** error)
{
GstSDPMessage *ret = NULL;
GString *bundled_mids = NULL;
gchar *bundle_ufrag = NULL;
gchar *bundle_pwd = NULL;
GArray *media_mapping = NULL;
GHashTable *all_mids =
g_hash_table_new_full (g_str_hash, g_str_equal, g_free, NULL);
GstSDPMessage *last_offer = _get_latest_self_generated_sdp (webrtc);
GList *seen_transceivers = NULL;
guint media_idx = 0;
int i;
gboolean no_more_mlines = FALSE;
gst_sdp_message_new (&ret);
gst_sdp_message_set_version (ret, "0");
{
gchar *v, *sess_id;
v = g_strdup_printf ("%u", webrtc->priv->offer_count++);
if (last_offer) {
const GstSDPOrigin *origin = gst_sdp_message_get_origin (last_offer);
sess_id = g_strdup (origin->sess_id);
} else {
sess_id = g_strdup_printf ("%" G_GUINT64_FORMAT, RANDOM_SESSION_ID);
}
gst_sdp_message_set_origin (ret, "-", sess_id, v, "IN", "IP4", "0.0.0.0");
g_free (sess_id);
g_free (v);
}
gst_sdp_message_set_session_name (ret, "-");
gst_sdp_message_add_time (ret, "0", "0", NULL);
gst_sdp_message_add_attribute (ret, "ice-options", "trickle");
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_BUNDLE) {
bundled_mids = g_string_new ("BUNDLE");
} else if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_MAX_COMPAT) {
bundled_mids = g_string_new ("BUNDLE");
}
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
GStrv last_bundle = NULL;
guint bundle_media_index;
media_mapping = gather_media_mapping (webrtc);
if (last_offer && _parse_bundle (last_offer, &last_bundle, NULL)
&& last_bundle && last_bundle[0]
&& _get_bundle_index (last_offer, last_bundle, &bundle_media_index)) {
bundle_ufrag =
g_strdup (_media_get_ice_ufrag (last_offer, bundle_media_index));
bundle_pwd =
g_strdup (_media_get_ice_pwd (last_offer, bundle_media_index));
} else {
_generate_ice_credentials (&bundle_ufrag, &bundle_pwd);
}
g_strfreev (last_bundle);
}
/* FIXME: recycle transceivers */
/* Fill up the renegotiated streams first */
if (last_offer) {
for (i = 0; i < gst_sdp_message_medias_len (last_offer); i++) {
GstWebRTCRTPTransceiver *trans = NULL;
const GstSDPMedia *last_media;
last_media = gst_sdp_message_get_media (last_offer, i);
if (g_strcmp0 (gst_sdp_media_get_media (last_media), "audio") == 0
|| g_strcmp0 (gst_sdp_media_get_media (last_media), "video") == 0) {
const gchar *last_mid;
int j;
last_mid = gst_sdp_media_get_attribute_val (last_media, "mid");
for (j = 0; j < webrtc->priv->transceivers->len; j++) {
WebRTCTransceiver *wtrans;
const gchar *mid;
trans = g_ptr_array_index (webrtc->priv->transceivers, j);
wtrans = WEBRTC_TRANSCEIVER (trans);
if (trans->mid)
mid = trans->mid;
else
mid = wtrans->pending_mid;
if (mid && g_strcmp0 (mid, last_mid) == 0) {
GstSDPMedia media;
memset (&media, 0, sizeof (media));
g_assert (!g_list_find (seen_transceivers, trans));
if (wtrans->mline_locked && trans->mline != media_idx) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Previous negotiatied transceiver <%s> with mid %s was in "
"mline %d but transceiver has locked mline %u",
GST_OBJECT_NAME (trans), trans->mid, media_idx, trans->mline);
goto cancel_offer;
}
GST_LOG_OBJECT (webrtc, "using previous negotiatied transceiver %"
GST_PTR_FORMAT " with mid %s into media index %u", trans,
trans->mid, media_idx);
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
media_mapping =
g_array_new (FALSE, FALSE,
sizeof (struct media_payload_map_item));
}
gst_sdp_media_init (&media);
if (!sdp_media_from_transceiver (webrtc, &media, last_media, trans,
media_idx, bundled_mids, 0, bundle_ufrag, bundle_pwd,
media_mapping, all_mids, &no_more_mlines, error)) {
gst_sdp_media_uninit (&media);
if (!*error)
g_set_error_literal (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Could not reuse transceiver");
}
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
g_array_free (media_mapping, TRUE);
media_mapping = NULL;
}
if (*error)
goto cancel_offer;
mid = gst_sdp_media_get_attribute_val (&media, "mid");
g_assert (mid && g_strcmp0 (last_mid, mid) == 0);
gst_sdp_message_add_media (ret, &media);
media_idx++;
gst_sdp_media_uninit (&media);
seen_transceivers = g_list_prepend (seen_transceivers, trans);
break;
}
}
} else if (g_strcmp0 (gst_sdp_media_get_media (last_media),
"application") == 0) {
GstSDPMedia media = { 0, };
gst_sdp_media_init (&media);
if (_add_data_channel_offer (webrtc, ret, &media, bundled_mids, 0,
bundle_ufrag, bundle_pwd, all_mids)) {
gst_sdp_message_add_media (ret, &media);
media_idx++;
} else {
gst_sdp_media_uninit (&media);
}
}
}
}
/* First, go over all transceivers and gather existing mids */
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *trans;
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
if (g_list_find (seen_transceivers, trans))
continue;
if (trans->mid) {
if (g_hash_table_contains (all_mids, trans->mid)) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Duplicate mid %s when creating offer", trans->mid);
goto cancel_offer;
}
g_hash_table_insert (all_mids, g_strdup (trans->mid), NULL);
} else if (WEBRTC_TRANSCEIVER (trans)->pending_mid &&
!g_hash_table_contains (all_mids,
WEBRTC_TRANSCEIVER (trans)->pending_mid)) {
g_hash_table_insert (all_mids,
g_strdup (WEBRTC_TRANSCEIVER (trans)->pending_mid), NULL);
}
}
/* add any extra streams */
for (;;) {
GstWebRTCRTPTransceiver *trans = NULL;
GstSDPMedia media = { 0, };
/* First find a transceiver requesting this m-line */
trans = _find_transceiver_for_mline (webrtc, media_idx);
if (trans) {
/* We can't have seen it already, because it is locked to this line,
* unless it's a no-more-mlines case
*/
if (!g_list_find (seen_transceivers, trans))
seen_transceivers = g_list_prepend (seen_transceivers, trans);
} else {
/* Otherwise find a free transceiver */
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
WebRTCTransceiver *wtrans;
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
wtrans = WEBRTC_TRANSCEIVER (trans);
/* don't add transceivers twice */
if (g_list_find (seen_transceivers, trans))
continue;
/* Ignore transceivers with a locked mline, as they would have been
* found above or will be used later */
if (wtrans->mline_locked)
continue;
seen_transceivers = g_list_prepend (seen_transceivers, trans);
/* don't add stopped transceivers */
if (trans->stopped) {
continue;
}
/* Otherwise take it */
break;
}
/* Stop if we got all transceivers */
if (i == webrtc->priv->transceivers->len) {
/* But try to add a data channel first, we do it here, because
* it can allow a locked m-line to be put after, so we need to
* do another iteration after.
*/
if (_message_get_datachannel_index (ret) == G_MAXUINT) {
GstSDPMedia media = { 0, };
gst_sdp_media_init (&media);
if (_add_data_channel_offer (webrtc, ret, &media, bundled_mids, 0,
bundle_ufrag, bundle_pwd, all_mids)) {
if (no_more_mlines) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Trying to add data channel but there is a"
" transceiver locked to line %d which doesn't have caps",
media_idx);
gst_sdp_media_uninit (&media);
goto cancel_offer;
}
gst_sdp_message_add_media (ret, &media);
media_idx++;
continue;
} else {
gst_sdp_media_uninit (&media);
}
}
/* Verify that we didn't ignore any locked m-line transceivers */
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
WebRTCTransceiver *wtrans;
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
wtrans = WEBRTC_TRANSCEIVER (trans);
/* don't add transceivers twice */
if (g_list_find (seen_transceivers, trans))
continue;
g_assert (wtrans->mline_locked);
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Tranceiver <%s> with mid %s has locked mline %d but the offer "
"only has %u sections", GST_OBJECT_NAME (trans), trans->mid,
trans->mline, media_idx);
goto cancel_offer;
}
break;
}
}
if (no_more_mlines) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Trying to add transceiver at line %u but there is a transceiver "
"with a locked mline for this line which doesn't have caps",
media_idx);
goto cancel_offer;
}
gst_sdp_media_init (&media);
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
media_mapping =
g_array_new (FALSE, FALSE, sizeof (struct media_payload_map_item));
}
GST_LOG_OBJECT (webrtc, "adding transceiver %" GST_PTR_FORMAT " at media "
"index %u", trans, media_idx);
if (sdp_media_from_transceiver (webrtc, &media, NULL, trans, media_idx,
bundled_mids, 0, bundle_ufrag, bundle_pwd, media_mapping, all_mids,
&no_more_mlines, error)) {
/* as per JSEP, a=rtcp-mux-only is only added for new streams */
gst_sdp_media_add_attribute (&media, "rtcp-mux-only", "");
gst_sdp_message_add_media (ret, &media);
media_idx++;
} else {
gst_sdp_media_uninit (&media);
}
if (webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE) {
g_array_free (media_mapping, TRUE);
media_mapping = NULL;
}
if (*error)
goto cancel_offer;
}
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
g_array_free (media_mapping, TRUE);
media_mapping = NULL;
}
webrtc->priv->max_sink_pad_serial = MAX (webrtc->priv->max_sink_pad_serial,
media_idx);
g_assert (media_idx == gst_sdp_message_medias_len (ret));
if (bundled_mids) {
gchar *mids = g_string_free (bundled_mids, FALSE);
gst_sdp_message_add_attribute (ret, "group", mids);
g_free (mids);
bundled_mids = NULL;
}
/* FIXME: pre-emptively setup receiving elements when needed */
if (webrtc->priv->last_generated_answer)
gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);
webrtc->priv->last_generated_answer = NULL;
if (webrtc->priv->last_generated_offer)
gst_webrtc_session_description_free (webrtc->priv->last_generated_offer);
{
GstSDPMessage *copy;
gst_sdp_message_copy (ret, &copy);
webrtc->priv->last_generated_offer =
gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_OFFER, copy);
}
out:
if (media_mapping)
g_array_free (media_mapping, TRUE);
g_hash_table_unref (all_mids);
g_list_free (seen_transceivers);
if (bundle_ufrag)
g_free (bundle_ufrag);
if (bundle_pwd)
g_free (bundle_pwd);
if (bundled_mids)
g_string_free (bundled_mids, TRUE);
return ret;
cancel_offer:
gst_sdp_message_free (ret);
ret = NULL;
goto out;
}
static void
_media_add_fec (GstSDPMedia * media, WebRTCTransceiver * trans, GstCaps * caps,
gint * rtx_target_pt)
{
guint i;
if (trans->fec_type == GST_WEBRTC_FEC_TYPE_NONE)
return;
for (i = 0; i < gst_caps_get_size (caps); i++) {
const GstStructure *s = gst_caps_get_structure (caps, i);
if (gst_structure_has_name (s, "application/x-rtp")) {
const gchar *encoding_name =
gst_structure_get_string (s, "encoding-name");
gint clock_rate;
gint pt;
if (gst_structure_get_int (s, "clock-rate", &clock_rate) &&
gst_structure_get_int (s, "payload", &pt)) {
if (!g_strcmp0 (encoding_name, "RED")) {
gchar *str;
str = g_strdup_printf ("%u", pt);
gst_sdp_media_add_format (media, str);
g_free (str);
str = g_strdup_printf ("%u red/%d", pt, clock_rate);
*rtx_target_pt = pt;
gst_sdp_media_add_attribute (media, "rtpmap", str);
g_free (str);
} else if (!g_strcmp0 (encoding_name, "ULPFEC")) {
gchar *str;
str = g_strdup_printf ("%u", pt);
gst_sdp_media_add_format (media, str);
g_free (str);
str = g_strdup_printf ("%u ulpfec/%d", pt, clock_rate);
gst_sdp_media_add_attribute (media, "rtpmap", str);
g_free (str);
}
}
}
}
}
static void
_media_add_rtx (GstSDPMedia * media, WebRTCTransceiver * trans,
GstCaps * offer_caps, gint target_pt, guint target_ssrc)
{
guint i;
const GstStructure *s;
if (trans->local_rtx_ssrc_map)
gst_structure_free (trans->local_rtx_ssrc_map);
trans->local_rtx_ssrc_map =
gst_structure_new_empty ("application/x-rtp-ssrc-map");
for (i = 0; i < gst_caps_get_size (offer_caps); i++) {
s = gst_caps_get_structure (offer_caps, i);
if (gst_structure_has_name (s, "application/x-rtp")) {
const gchar *encoding_name =
gst_structure_get_string (s, "encoding-name");
const gchar *apt_str = gst_structure_get_string (s, "apt");
gint apt;
gint clock_rate;
gint pt;
if (!apt_str)
continue;
apt = atoi (apt_str);
if (gst_structure_get_int (s, "clock-rate", &clock_rate) &&
gst_structure_get_int (s, "payload", &pt) && apt == target_pt) {
if (!g_strcmp0 (encoding_name, "RTX")) {
gchar *str;
str = g_strdup_printf ("%u", pt);
gst_sdp_media_add_format (media, str);
g_free (str);
str = g_strdup_printf ("%u rtx/%d", pt, clock_rate);
gst_sdp_media_add_attribute (media, "rtpmap", str);
g_free (str);
str = g_strdup_printf ("%d apt=%d", pt, apt);
gst_sdp_media_add_attribute (media, "fmtp", str);
g_free (str);
str = g_strdup_printf ("%u", target_ssrc);
gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT,
g_random_int (), NULL);
g_free (str);
}
}
}
}
}
static gboolean
_update_transceiver_kind_from_caps (GstWebRTCRTPTransceiver * trans,
const GstCaps * caps)
{
GstWebRTCKind kind = webrtc_kind_from_caps (caps);
if (trans->kind == kind)
return TRUE;
if (trans->kind == GST_WEBRTC_KIND_UNKNOWN) {
trans->kind = kind;
return TRUE;
} else {
return FALSE;
}
}
static void
_get_rtx_target_pt_and_ssrc_from_caps (GstCaps * answer_caps, gint * target_pt,
guint * target_ssrc)
{
const GstStructure *s = gst_caps_get_structure (answer_caps, 0);
gst_structure_get_int (s, "payload", target_pt);
gst_structure_get_uint (s, "ssrc", target_ssrc);
}
/* TODO: use the options argument */
static GstSDPMessage *
_create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
GError ** error)
{
GstSDPMessage *ret = NULL;
const GstWebRTCSessionDescription *pending_remote =
webrtc->pending_remote_description;
guint i;
GStrv bundled = NULL;
guint bundle_idx = 0;
GString *bundled_mids = NULL;
gchar *bundle_ufrag = NULL;
gchar *bundle_pwd = NULL;
GList *seen_transceivers = NULL;
GstSDPMessage *last_answer = _get_latest_self_generated_sdp (webrtc);
if (!webrtc->pending_remote_description) {
g_set_error_literal (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INVALID_STATE,
"Asked to create an answer without a remote description");
return NULL;
}
if (!_parse_bundle (pending_remote->sdp, &bundled, error))
goto out;
if (bundled) {
GStrv last_bundle = NULL;
guint bundle_media_index;
if (!_get_bundle_index (pending_remote->sdp, bundled, &bundle_idx)) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Bundle tag is %s but no media found matching", bundled[0]);
goto out;
}
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE) {
bundled_mids = g_string_new ("BUNDLE");
}
if (last_answer && _parse_bundle (last_answer, &last_bundle, NULL)
&& last_bundle && last_bundle[0]
&& _get_bundle_index (last_answer, last_bundle, &bundle_media_index)) {
bundle_ufrag =
g_strdup (_media_get_ice_ufrag (last_answer, bundle_media_index));
bundle_pwd =
g_strdup (_media_get_ice_pwd (last_answer, bundle_media_index));
} else {
_generate_ice_credentials (&bundle_ufrag, &bundle_pwd);
}
g_strfreev (last_bundle);
}
gst_sdp_message_new (&ret);
gst_sdp_message_set_version (ret, "0");
{
const GstSDPOrigin *offer_origin =
gst_sdp_message_get_origin (pending_remote->sdp);
gst_sdp_message_set_origin (ret, "-", offer_origin->sess_id,
offer_origin->sess_version, "IN", "IP4", "0.0.0.0");
}
gst_sdp_message_set_session_name (ret, "-");
for (i = 0; i < gst_sdp_message_attributes_len (pending_remote->sdp); i++) {
const GstSDPAttribute *attr =
gst_sdp_message_get_attribute (pending_remote->sdp, i);
if (g_strcmp0 (attr->key, "ice-options") == 0) {
gst_sdp_message_add_attribute (ret, attr->key, attr->value);
}
}
for (i = 0; i < gst_sdp_message_medias_len (pending_remote->sdp); i++) {
GstSDPMedia *media = NULL;
GstSDPMedia *offer_media;
GstWebRTCDTLSSetup offer_setup, answer_setup;
guint j, k;
gboolean bundle_only;
const gchar *mid;
offer_media =
(GstSDPMedia *) gst_sdp_message_get_media (pending_remote->sdp, i);
bundle_only = _media_has_attribute_key (offer_media, "bundle-only");
gst_sdp_media_new (&media);
if (bundle_only && webrtc->bundle_policy == GST_WEBRTC_BUNDLE_POLICY_NONE)
gst_sdp_media_set_port_info (media, 0, 0);
else
gst_sdp_media_set_port_info (media, 9, 0);
gst_sdp_media_add_connection (media, "IN", "IP4", "0.0.0.0", 0, 0);
{
gchar *ufrag, *pwd;
/* FIXME: deal with ICE restarts */
if (last_answer && i < gst_sdp_message_medias_len (last_answer)) {
ufrag = g_strdup (_media_get_ice_ufrag (last_answer, i));
pwd = g_strdup (_media_get_ice_pwd (last_answer, i));
} else {
if (!bundled) {
_generate_ice_credentials (&ufrag, &pwd);
} else {
ufrag = g_strdup (bundle_ufrag);
pwd = g_strdup (bundle_pwd);
}
}
gst_sdp_media_add_attribute (media, "ice-ufrag", ufrag);
gst_sdp_media_add_attribute (media, "ice-pwd", pwd);
g_free (ufrag);
g_free (pwd);
}
for (j = 0; j < gst_sdp_media_attributes_len (offer_media); j++) {
const GstSDPAttribute *attr =
gst_sdp_media_get_attribute (offer_media, j);
if (g_strcmp0 (attr->key, "mid") == 0
|| g_strcmp0 (attr->key, "rtcp-mux") == 0) {
gst_sdp_media_add_attribute (media, attr->key, attr->value);
/* FIXME: handle anything we want to keep */
}
}
mid = gst_sdp_media_get_attribute_val (media, "mid");
/* XXX: not strictly required but a lot of functionality requires a mid */
g_assert (mid);
/* set the a=setup: attribute */
offer_setup = _get_dtls_setup_from_media (offer_media);
answer_setup = _intersect_dtls_setup (offer_setup);
if (answer_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
GST_WARNING_OBJECT (webrtc, "Could not intersect offer setup with "
"transceiver direction");
goto rejected;
}
_media_replace_setup (media, answer_setup);
if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "application") == 0) {
int sctp_port;
if (gst_sdp_media_formats_len (offer_media) != 1) {
GST_WARNING_OBJECT (webrtc, "Could not find a format in the m= line "
"for webrtc-datachannel");
goto rejected;
}
sctp_port = _get_sctp_port_from_media (offer_media);
if (sctp_port == -1) {
GST_WARNING_OBJECT (webrtc, "media does not contain a sctp port");
goto rejected;
}
/* XXX: older browsers will produce a different SDP format for data
* channel that is currently not parsed correctly */
gst_sdp_media_set_proto (media, "UDP/DTLS/SCTP");
gst_sdp_media_set_media (media, "application");
gst_sdp_media_set_port_info (media, 9, 0);
gst_sdp_media_add_format (media, "webrtc-datachannel");
/* FIXME: negotiate this properly on renegotiation */
gst_sdp_media_add_attribute (media, "sctp-port", "5000");
_get_or_create_data_channel_transports (webrtc,
bundled_mids ? bundle_idx : i);
if (bundled_mids) {
g_assert (mid);
g_string_append_printf (bundled_mids, " %s", mid);
}
_add_fingerprint_to_media (webrtc->priv->sctp_transport->transport,
media);
} else if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "audio") == 0
|| g_strcmp0 (gst_sdp_media_get_media (offer_media), "video") == 0) {
GstCaps *offer_caps, *answer_caps = NULL;
GstWebRTCRTPTransceiver *rtp_trans = NULL;
WebRTCTransceiver *trans = NULL;
GstWebRTCRTPTransceiverDirection offer_dir, answer_dir;
gint target_pt = -1;
gint original_target_pt = -1;
guint target_ssrc = 0;
gst_sdp_media_set_proto (media, "UDP/TLS/RTP/SAVPF");
offer_caps = _rtp_caps_from_media (offer_media);
if (last_answer && i < gst_sdp_message_medias_len (last_answer)
&& (rtp_trans = _find_transceiver_for_mid (webrtc, mid))) {
const GstSDPMedia *last_media =
gst_sdp_message_get_media (last_answer, i);
const gchar *last_mid =
gst_sdp_media_get_attribute_val (last_media, "mid");
GstCaps *current_caps;
/* FIXME: assumes no shenanigans with recycling transceivers */
g_assert (g_strcmp0 (mid, last_mid) == 0);
current_caps = _find_codec_preferences (webrtc, rtp_trans, i, error);
if (*error) {
gst_caps_unref (offer_caps);
goto rejected;
}
if (!current_caps)
current_caps = _rtp_caps_from_media (last_media);
if (current_caps) {
answer_caps = gst_caps_intersect (offer_caps, current_caps);
if (gst_caps_is_empty (answer_caps)) {
GST_WARNING_OBJECT (webrtc, "Caps from offer for m-line %d (%"
GST_PTR_FORMAT ") don't intersect with caps from codec"
" preferences and transceiver %" GST_PTR_FORMAT, i, offer_caps,
current_caps);
gst_caps_unref (current_caps);
gst_caps_unref (answer_caps);
gst_caps_unref (offer_caps);
goto rejected;
}
gst_caps_unref (current_caps);
}
/* XXX: In theory we're meant to use the sendrecv formats for the
* inactive direction however we don't know what that may be and would
* require asking outside what it expects to possibly send later */
GST_LOG_OBJECT (webrtc, "Found existing previously negotiated "
"transceiver %" GST_PTR_FORMAT " from mid %s for mline %u "
"using caps %" GST_PTR_FORMAT, rtp_trans, mid, i, answer_caps);
} else {
for (j = 0; j < webrtc->priv->transceivers->len; j++) {
GstCaps *trans_caps;
rtp_trans = g_ptr_array_index (webrtc->priv->transceivers, j);
if (g_list_find (seen_transceivers, rtp_trans)) {
/* Don't double allocate a transceiver to multiple mlines */
rtp_trans = NULL;
continue;
}
trans_caps = _find_codec_preferences (webrtc, rtp_trans, j, error);
if (*error) {
gst_caps_unref (offer_caps);
goto rejected;
}
GST_LOG_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT
" and %" GST_PTR_FORMAT, offer_caps, trans_caps);
/* FIXME: technically this is a little overreaching as some fields we
* we can deal with not having and/or we may have unrecognized fields
* that we cannot actually support */
if (trans_caps) {
answer_caps = gst_caps_intersect (offer_caps, trans_caps);
gst_caps_unref (trans_caps);
if (answer_caps) {
if (!gst_caps_is_empty (answer_caps)) {
GST_LOG_OBJECT (webrtc,
"found compatible transceiver %" GST_PTR_FORMAT
" for offer media %u", rtp_trans, i);
break;
}
gst_caps_unref (answer_caps);
answer_caps = NULL;
}
}
rtp_trans = NULL;
}
}
if (rtp_trans) {
answer_dir = rtp_trans->direction;
g_assert (answer_caps != NULL);
} else {
/* if no transceiver, then we only receive that stream and respond with
* the intersection with the transceivers codec preferences caps */
answer_dir = GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY;
GST_WARNING_OBJECT (webrtc, "did not find compatible transceiver for "
"offer caps %" GST_PTR_FORMAT ", will only receive", offer_caps);
}
if (!rtp_trans) {
GstCaps *trans_caps;
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
if (g_strcmp0 (gst_sdp_media_get_media (offer_media), "audio") == 0)
kind = GST_WEBRTC_KIND_AUDIO;
else if (g_strcmp0 (gst_sdp_media_get_media (offer_media),
"video") == 0)
kind = GST_WEBRTC_KIND_VIDEO;
else
GST_LOG_OBJECT (webrtc, "Unknown media kind %s",
GST_STR_NULL (gst_sdp_media_get_media (offer_media)));
trans = _create_webrtc_transceiver (webrtc, answer_dir, i, kind, NULL);
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
PC_UNLOCK (webrtc);
g_signal_emit (webrtc,
gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0, rtp_trans);
PC_LOCK (webrtc);
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT
" for mline %u with media kind %d", trans, i, kind);
trans_caps = _find_codec_preferences (webrtc, rtp_trans, i, error);
if (*error) {
gst_caps_unref (offer_caps);
goto rejected;
}
GST_TRACE_OBJECT (webrtc, "trying to compare %" GST_PTR_FORMAT
" and %" GST_PTR_FORMAT, offer_caps, trans_caps);
/* FIXME: technically this is a little overreaching as some fields we
* we can deal with not having and/or we may have unrecognized fields
* that we cannot actually support */
if (trans_caps) {
answer_caps = gst_caps_intersect (offer_caps, trans_caps);
gst_clear_caps (&trans_caps);
} else {
answer_caps = gst_caps_ref (offer_caps);
}
} else {
trans = WEBRTC_TRANSCEIVER (rtp_trans);
}
seen_transceivers = g_list_prepend (seen_transceivers, rtp_trans);
if (gst_caps_is_empty (answer_caps)) {
GST_WARNING_OBJECT (webrtc, "Could not create caps for media");
gst_clear_caps (&answer_caps);
gst_clear_caps (&offer_caps);
goto rejected;
}
if (!_update_transceiver_kind_from_caps (rtp_trans, answer_caps)) {
GstWebRTCKind caps_kind = webrtc_kind_from_caps (answer_caps);
GST_WARNING_OBJECT (webrtc,
"Trying to change kind of transceiver %" GST_PTR_FORMAT
" at m-line %d from %s (%d) to %s (%d)", trans, rtp_trans->mline,
gst_webrtc_kind_to_string (rtp_trans->kind), rtp_trans->kind,
gst_webrtc_kind_to_string (caps_kind), caps_kind);
}
answer_caps = gst_caps_make_writable (answer_caps);
for (k = 0; k < gst_caps_get_size (answer_caps); k++) {
GstStructure *s = gst_caps_get_structure (answer_caps, k);
/* taken from the offer sdp already and already intersected above */
gst_structure_remove_field (s, "a-mid");
if (!trans->do_nack)
gst_structure_remove_fields (s, "rtcp-fb-nack", NULL);
}
if (gst_sdp_media_set_media_from_caps (answer_caps, media) != GST_SDP_OK) {
GST_WARNING_OBJECT (webrtc,
"Could not build media from caps %" GST_PTR_FORMAT, answer_caps);
gst_clear_caps (&answer_caps);
gst_clear_caps (&offer_caps);
goto rejected;
}
_get_rtx_target_pt_and_ssrc_from_caps (answer_caps, &target_pt,
&target_ssrc);
original_target_pt = target_pt;
_media_add_fec (media, trans, offer_caps, &target_pt);
if (trans->do_nack) {
_media_add_rtx (media, trans, offer_caps, target_pt, target_ssrc);
if (target_pt != original_target_pt)
_media_add_rtx (media, trans, offer_caps, original_target_pt,
target_ssrc);
}
if (answer_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY)
_media_add_ssrcs (media, answer_caps, webrtc,
WEBRTC_TRANSCEIVER (rtp_trans));
gst_caps_unref (answer_caps);
answer_caps = NULL;
/* set the new media direction */
offer_dir = _get_direction_from_media (offer_media);
answer_dir = _intersect_answer_directions (offer_dir, answer_dir);
if (answer_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) {
GST_WARNING_OBJECT (webrtc, "Could not intersect offer direction with "
"transceiver direction");
gst_caps_unref (offer_caps);
goto rejected;
}
_media_replace_direction (media, answer_dir);
if (!trans->stream) {
TransportStream *item;
item =
_get_or_create_transport_stream (webrtc,
bundled_mids ? bundle_idx : i, FALSE);
webrtc_transceiver_set_transport (trans, item);
}
if (bundled_mids) {
const gchar *mid = gst_sdp_media_get_attribute_val (media, "mid");
g_assert (mid);
g_string_append_printf (bundled_mids, " %s", mid);
}
/* set the a=fingerprint: for this transport */
_add_fingerprint_to_media (trans->stream->transport, media);
gst_caps_unref (offer_caps);
} else {
GST_WARNING_OBJECT (webrtc, "unknown m= line media name");
goto rejected;
}
if (0) {
rejected:
if (error && *error)
GST_INFO_OBJECT (webrtc, "media %u rejected: %s", i, (*error)->message);
else
GST_INFO_OBJECT (webrtc, "media %u rejected", i);
gst_sdp_media_free (media);
gst_sdp_media_copy (offer_media, &media);
gst_sdp_media_set_port_info (media, 0, 0);
/* Clear error here as it is not propagated to the caller and the media
* is just skipped, i.e. more iterations are going to happen. */
g_clear_error (error);
}
gst_sdp_message_add_media (ret, media);
gst_sdp_media_free (media);
}
if (bundled_mids) {
gchar *mids = g_string_free (bundled_mids, FALSE);
gst_sdp_message_add_attribute (ret, "group", mids);
g_free (mids);
}
if (bundle_ufrag)
g_free (bundle_ufrag);
if (bundle_pwd)
g_free (bundle_pwd);
/* FIXME: can we add not matched transceivers? */
/* XXX: only true for the initial offerer */
gst_webrtc_ice_set_is_controller (webrtc->priv->ice, FALSE);
out:
g_strfreev (bundled);
g_list_free (seen_transceivers);
if (webrtc->priv->last_generated_offer)
gst_webrtc_session_description_free (webrtc->priv->last_generated_offer);
webrtc->priv->last_generated_offer = NULL;
if (webrtc->priv->last_generated_answer)
gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);
{
GstSDPMessage *copy;
gst_sdp_message_copy (ret, &copy);
webrtc->priv->last_generated_answer =
gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER, copy);
}
return ret;
}
struct create_sdp
{
GstStructure *options;
GstWebRTCSDPType type;
};
static GstStructure *
_create_sdp_task (GstWebRTCBin * webrtc, struct create_sdp *data)
{
GstWebRTCSessionDescription *desc = NULL;
GstSDPMessage *sdp = NULL;
GstStructure *s = NULL;
GError *error = NULL;
GST_INFO_OBJECT (webrtc, "creating %s sdp with options %" GST_PTR_FORMAT,
gst_webrtc_sdp_type_to_string (data->type), data->options);
if (data->type == GST_WEBRTC_SDP_TYPE_OFFER)
sdp = _create_offer_task (webrtc, data->options, &error);
else if (data->type == GST_WEBRTC_SDP_TYPE_ANSWER)
sdp = _create_answer_task (webrtc, data->options, &error);
else {
g_assert_not_reached ();
goto out;
}
if (sdp) {
desc = gst_webrtc_session_description_new (data->type, sdp);
s = gst_structure_new ("application/x-gst-promise",
gst_webrtc_sdp_type_to_string (data->type),
GST_TYPE_WEBRTC_SESSION_DESCRIPTION, desc, NULL);
} else {
g_warn_if_fail (error != NULL);
GST_WARNING_OBJECT (webrtc, "returning error: %s",
error ? error->message : "Unknown");
s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
g_clear_error (&error);
}
out:
if (desc)
gst_webrtc_session_description_free (desc);
return s;
}
static void
_free_create_sdp_data (struct create_sdp *data)
{
if (data->options)
gst_structure_free (data->options);
g_free (data);
}
static void
gst_webrtc_bin_create_offer (GstWebRTCBin * webrtc,
const GstStructure * options, GstPromise * promise)
{
struct create_sdp *data = g_new0 (struct create_sdp, 1);
if (options)
data->options = gst_structure_copy (options);
data->type = GST_WEBRTC_SDP_TYPE_OFFER;
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
data, (GDestroyNotify) _free_create_sdp_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not create offer. webrtcbin is closed");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
gst_promise_reply (promise, s);
g_clear_error (&error);
}
}
static void
gst_webrtc_bin_create_answer (GstWebRTCBin * webrtc,
const GstStructure * options, GstPromise * promise)
{
struct create_sdp *data = g_new0 (struct create_sdp, 1);
if (options)
data->options = gst_structure_copy (options);
data->type = GST_WEBRTC_SDP_TYPE_ANSWER;
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _create_sdp_task,
data, (GDestroyNotify) _free_create_sdp_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not create answer. webrtcbin is closed.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
gst_promise_reply (promise, s);
g_clear_error (&error);
}
}
static GstWebRTCBinPad *
_create_pad_for_sdp_media (GstWebRTCBin * webrtc, GstPadDirection direction,
GstWebRTCRTPTransceiver * trans, guint serial, char *msid)
{
GstWebRTCBinPad *pad;
gchar *pad_name;
if (direction == GST_PAD_SINK) {
if (serial == G_MAXUINT)
serial = webrtc->priv->max_sink_pad_serial++;
} else {
serial = webrtc->priv->src_pad_counter++;
}
pad_name =
g_strdup_printf ("%s_%u", direction == GST_PAD_SRC ? "src" : "sink",
serial);
pad = gst_webrtc_bin_pad_new (pad_name, direction, msid);
g_free (pad_name);
pad->trans = gst_object_ref (trans);
return pad;
}
static GstWebRTCRTPTransceiver *
_find_transceiver_for_sdp_media (GstWebRTCBin * webrtc,
const GstSDPMessage * sdp, guint media_idx)
{
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
GstWebRTCRTPTransceiver *ret = NULL;
int i;
for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i);
if (g_strcmp0 (attr->key, "mid") == 0) {
if ((ret = _find_transceiver_for_mid (webrtc, attr->value)))
goto out;
}
}
ret = _find_transceiver (webrtc, &media_idx,
(FindTransceiverFunc) transceiver_match_for_mline);
out:
GST_TRACE_OBJECT (webrtc, "Found transceiver %" GST_PTR_FORMAT, ret);
return ret;
}
static GstElement *
_build_fec_encoder (GstWebRTCBin * webrtc, WebRTCTransceiver * trans)
{
GstWebRTCRTPTransceiver *rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
guint ulpfec_pt = 0, red_pt = 0;
GstPad *sinkpad, *srcpad, *ghost;
GstElement *ret;
if (trans->stream) {
ulpfec_pt =
transport_stream_get_pt (trans->stream, "ULPFEC", rtp_trans->mline);
red_pt = transport_stream_get_pt (trans->stream, "RED", rtp_trans->mline);
}
if (trans->ulpfecenc || trans->redenc) {
g_critical ("webrtcbin: duplicate call to create a fec encoder or "
"red encoder!");
return NULL;
}
GST_DEBUG_OBJECT (webrtc,
"Creating ULPFEC encoder for mline %u with pt %d", rtp_trans->mline,
ulpfec_pt);
ret = gst_bin_new (NULL);
trans->ulpfecenc = gst_element_factory_make ("rtpulpfecenc", NULL);
gst_object_ref_sink (trans->ulpfecenc);
if (!gst_bin_add (GST_BIN (ret), trans->ulpfecenc))
g_warn_if_reached ();
sinkpad = gst_element_get_static_pad (trans->ulpfecenc, "sink");
g_object_bind_property (rtp_trans, "fec-percentage", trans->ulpfecenc,
"percentage", G_BINDING_DEFAULT);
trans->redenc = gst_element_factory_make ("rtpredenc", NULL);
gst_object_ref_sink (trans->redenc);
GST_DEBUG_OBJECT (webrtc, "Creating RED encoder for mline %u with pt %d",
rtp_trans->mline, red_pt);
gst_bin_add (GST_BIN (ret), trans->redenc);
gst_element_link (trans->ulpfecenc, trans->redenc);
ghost = gst_ghost_pad_new ("sink", sinkpad);
gst_clear_object (&sinkpad);
gst_element_add_pad (ret, ghost);
ghost = NULL;
srcpad = gst_element_get_static_pad (trans->redenc, "src");
ghost = gst_ghost_pad_new ("src", srcpad);
gst_clear_object (&srcpad);
gst_element_add_pad (ret, ghost);
ghost = NULL;
return ret;
}
static gboolean
_merge_structure (GQuark field_id, const GValue * value, gpointer user_data)
{
GstStructure *s = user_data;
gst_structure_id_set_value (s, field_id, value);
return TRUE;
}
#define GST_WEBRTC_PAYLOAD_TYPE "gst.webrtcbin.payload.type"
static void
try_match_transceiver_with_fec_decoder (GstWebRTCBin * webrtc,
WebRTCTransceiver * trans)
{
GList *l;
for (l = trans->stream->fecdecs; l; l = l->next) {
GstElement *fecdec = GST_ELEMENT (l->data);
gboolean found_transceiver = FALSE;
int original_pt;
guint i;
original_pt =
GPOINTER_TO_INT (g_object_get_data (G_OBJECT (fecdec),
GST_WEBRTC_PAYLOAD_TYPE));
if (original_pt <= 0) {
GST_WARNING_OBJECT (trans, "failed to match fec decoder with "
"transceiver, fec decoder %" GST_PTR_FORMAT " does not contain a "
"valid payload type", fecdec);
continue;
}
for (i = 0; i < trans->stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (trans->stream->ptmap, PtMapItem, i);
/* FIXME: this only works for a 1-1 original_pt->fec_pt mapping */
if (original_pt == item->pt && item->media_idx != -1
&& item->media_idx == trans->parent.mline) {
if (trans->ulpfecdec) {
GST_FIXME_OBJECT (trans, "cannot");
gst_clear_object (&trans->ulpfecdec);
}
trans->ulpfecdec = gst_object_ref (fecdec);
found_transceiver = TRUE;
break;
}
}
if (!found_transceiver) {
GST_WARNING_OBJECT (trans, "failed to match fec decoder with "
"transceiver");
}
}
}
static void
_set_internal_rtpbin_element_props_from_stream (GstWebRTCBin * webrtc,
TransportStream * stream)
{
GstStructure *merged_local_rtx_ssrc_map;
GstStructure *pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
GValue red_pt_array = { 0, };
gint *rtx_pt;
gsize rtx_count;
gsize i;
gst_value_array_init (&red_pt_array, 0);
rtx_pt = transport_stream_get_all_pt (stream, "RTX", &rtx_count);
GST_DEBUG_OBJECT (stream, "have %" G_GSIZE_FORMAT " rtx payloads", rtx_count);
for (i = 0; i < rtx_count; i++) {
GstCaps *rtx_caps = transport_stream_get_caps_for_pt (stream, rtx_pt[i]);
const GstStructure *s = gst_caps_get_structure (rtx_caps, 0);
const gchar *apt = gst_structure_get_string (s, "apt");
GST_LOG_OBJECT (stream, "setting rtx mapping: %s -> %u", apt, rtx_pt[i]);
gst_structure_set (pt_map, apt, G_TYPE_UINT, rtx_pt[i], NULL);
}
GST_DEBUG_OBJECT (stream, "setting payload map on %" GST_PTR_FORMAT " : %"
GST_PTR_FORMAT " and %" GST_PTR_FORMAT, stream->rtxreceive,
stream->rtxsend, pt_map);
if (stream->rtxreceive)
g_object_set (stream->rtxreceive, "payload-type-map", pt_map, NULL);
if (stream->rtxsend)
g_object_set (stream->rtxsend, "payload-type-map", pt_map, NULL);
gst_structure_free (pt_map);
g_clear_pointer (&rtx_pt, g_free);
merged_local_rtx_ssrc_map =
gst_structure_new_empty ("application/x-rtp-ssrc-map");
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *rtp_trans =
g_ptr_array_index (webrtc->priv->transceivers, i);
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
if (trans->stream == stream) {
gint ulpfec_pt, red_pt = 0;
ulpfec_pt = transport_stream_get_pt (stream, "ULPFEC", rtp_trans->mline);
if (ulpfec_pt <= 0)
ulpfec_pt = 0;
red_pt = transport_stream_get_pt (stream, "RED", rtp_trans->mline);
if (red_pt <= 0) {
red_pt = -1;
} else {
GValue ptval = { 0, };
g_value_init (&ptval, G_TYPE_INT);
g_value_set_int (&ptval, red_pt);
gst_value_array_append_value (&red_pt_array, &ptval);
g_value_unset (&ptval);
}
GST_DEBUG_OBJECT (webrtc, "stream %" GST_PTR_FORMAT " transceiver %"
GST_PTR_FORMAT " has FEC payload %d and RED payload %d", stream,
trans, ulpfec_pt, red_pt);
if (trans->ulpfecenc) {
guint ulpfecenc_pt = ulpfec_pt;
if (ulpfecenc_pt == 0)
ulpfecenc_pt = 255;
g_object_set (trans->ulpfecenc, "pt", ulpfecenc_pt, "multipacket",
rtp_trans->kind == GST_WEBRTC_KIND_VIDEO, "percentage",
trans->fec_percentage, NULL);
}
try_match_transceiver_with_fec_decoder (webrtc, trans);
if (trans->ulpfecdec) {
g_object_set (trans->ulpfecdec, "passthrough", ulpfec_pt == 0, "pt",
ulpfec_pt, NULL);
}
if (trans->redenc) {
gboolean always_produce = TRUE;
if (red_pt == -1) {
/* passthrough settings */
red_pt = 0;
always_produce = FALSE;
}
g_object_set (trans->redenc, "pt", red_pt, "allow-no-red-blocks",
always_produce, NULL);
}
if (trans->local_rtx_ssrc_map) {
gst_structure_foreach (trans->local_rtx_ssrc_map,
_merge_structure, merged_local_rtx_ssrc_map);
}
}
}
if (stream->rtxsend)
g_object_set (stream->rtxsend, "ssrc-map", merged_local_rtx_ssrc_map, NULL);
gst_clear_structure (&merged_local_rtx_ssrc_map);
if (stream->reddec) {
g_object_set_property (G_OBJECT (stream->reddec), "payloads",
&red_pt_array);
}
g_value_unset (&red_pt_array);
}
static GstPad *
_connect_input_stream (GstWebRTCBin * webrtc, GstWebRTCBinPad * pad)
{
/*
* Not-bundle case:
*
* ,--------------------------------------------webrtcbin--------------------------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ,---clocksync---, ; ; ; ; ;
* ; ; ; ; send_rtcp_src_%u o---o rtcp_sink ; ;
* ; sink_%u ; ; ,---fec encoder---, ; ; '---------------------' ;
* o---------o sink src o-o sink src o--o send_rtp_sink_%u ; ;
* ; '---------------' ,-----------------, '--------------------' ;
* '-------------------------------------------------------------------------------------------------'
*/
/*
* Bundle case:
* ,-----------------------------------------------------webrtcbin---------------------------------------------------,
* ; ;
* ; ,-------rtpbin-------, ,--transport_send_%u--, ;
* ; ; send_rtp_src_%u o---o rtp_sink ; ;
* ; ; ; ; ; ;
* ; sink_%u ,---clocksync---, ,---fec encoder---, ,---funnel---, ; send_rtcp_src_%u o---o rtcp_sink ; ;
* o----------o sink src o-o sink src o--o sink_%u ; ; ; '---------------------' ;
* ; '---------------' ,-----------------, ; ; ; ; ;
* ; ; src o-o send_rtp_sink_%u ; ;
* ; sink_%u ,---clocksync---, ,---fec encoder---, ; ; ; ; ;
* o----------o sink src o-o sink src o--o sink%u ; '--------------------' ;
* ; '---------------' ,-----------------, '------------' ;
* '-----------------------------------------------------------------------------------------------------------------'
*/
GstPadTemplate *rtp_templ;
GstPad *rtp_sink, *sinkpad, *srcpad;
gchar *pad_name;
WebRTCTransceiver *trans;
GstElement *clocksync;
GstElement *fec_encoder;
g_return_val_if_fail (pad->trans != NULL, NULL);
trans = WEBRTC_TRANSCEIVER (pad->trans);
GST_INFO_OBJECT (pad, "linking input stream %u", pad->trans->mline);
g_assert (trans->stream);
clocksync = gst_element_factory_make ("clocksync", NULL);
g_object_set (clocksync, "sync", TRUE, NULL);
gst_bin_add (GST_BIN (webrtc), clocksync);
gst_element_sync_state_with_parent (clocksync);
srcpad = gst_element_get_static_pad (clocksync, "src");
fec_encoder = _build_fec_encoder (webrtc, trans);
if (!fec_encoder) {
g_warn_if_reached ();
return NULL;
}
_set_internal_rtpbin_element_props_from_stream (webrtc, trans->stream);
gst_bin_add (GST_BIN (webrtc), fec_encoder);
gst_element_sync_state_with_parent (fec_encoder);
sinkpad = gst_element_get_static_pad (fec_encoder, "sink");
if (gst_pad_link (srcpad, sinkpad) != GST_PAD_LINK_OK)
g_warn_if_reached ();
gst_clear_object (&srcpad);
gst_clear_object (&sinkpad);
sinkpad = gst_element_get_static_pad (clocksync, "sink");
srcpad = gst_element_get_static_pad (fec_encoder, "src");
if (!webrtc->rtpfunnel) {
rtp_templ =
_find_pad_template (webrtc->rtpbin, GST_PAD_SINK, GST_PAD_REQUEST,
"send_rtp_sink_%u");
g_assert (rtp_templ);
pad_name = g_strdup_printf ("send_rtp_sink_%u", pad->trans->mline);
rtp_sink =
gst_element_request_pad (webrtc->rtpbin, rtp_templ, pad_name, NULL);
g_free (pad_name);
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);
} else {
gchar *pad_name = g_strdup_printf ("sink_%u", pad->trans->mline);
GstPad *funnel_sinkpad =
gst_element_request_pad_simple (webrtc->rtpfunnel, pad_name);
gst_pad_link (srcpad, funnel_sinkpad);
g_free (pad_name);
gst_object_unref (funnel_sinkpad);
}
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), sinkpad);
gst_clear_object (&srcpad);
gst_clear_object (&sinkpad);
gst_element_sync_state_with_parent (GST_ELEMENT (trans->stream->send_bin));
return GST_PAD (pad);
}
/* output pads are receiving elements */
static void
_connect_output_stream (GstWebRTCBin * webrtc,
TransportStream * stream, guint session_id)
{
/*
* ,------------------------webrtcbin------------------------,
* ; ,---------rtpbin---------, ;
* ; ,-transport_receive_%u--, ; ; ;
* ; ; rtp_src o---o recv_rtp_sink_%u ; ;
* ; ; ; ; ; ;
* ; ; rtcp_src o---o recv_rtcp_sink_%u ; ;
* ; '-----------------------' ; ; ; src_%u
* ; ; recv_rtp_src_%u_%u_%u o--o
* ; '------------------------' ;
* '---------------------------------------------------------'
*/
gchar *pad_name;
if (stream->output_connected) {
GST_DEBUG_OBJECT (webrtc, "stream %" GST_PTR_FORMAT " is already "
"connected to rtpbin. Not connecting", stream);
return;
}
GST_INFO_OBJECT (webrtc, "linking output stream %u %" GST_PTR_FORMAT,
session_id, stream);
pad_name = g_strdup_printf ("recv_rtp_sink_%u", session_id);
if (!gst_element_link_pads (GST_ELEMENT (stream->receive_bin),
"rtp_src", GST_ELEMENT (webrtc->rtpbin), pad_name))
g_warn_if_reached ();
g_free (pad_name);
gst_element_sync_state_with_parent (GST_ELEMENT (stream->receive_bin));
/* The webrtcbin src_%u output pads will be created when rtpbin receives
* data on that stream in on_rtpbin_pad_added() */
stream->output_connected = TRUE;
}
typedef struct
{
guint mlineindex;
gchar *candidate;
GstPromise *promise;
} IceCandidateItem;
static void
_clear_ice_candidate_item (IceCandidateItem * item)
{
g_free (item->candidate);
if (item->promise)
gst_promise_unref (item->promise);
}
static void
_add_ice_candidate (GstWebRTCBin * webrtc, IceCandidateItem * item,
gboolean drop_invalid)
{
GstWebRTCICEStream *stream;
stream = _find_ice_stream_for_session (webrtc, item->mlineindex);
if (stream == NULL) {
if (drop_invalid) {
if (item->promise) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"Unknown mline %u, dropping", item->mlineindex);
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
gst_promise_reply (item->promise, s);
g_clear_error (&error);
} else {
GST_WARNING_OBJECT (webrtc, "Unknown mline %u, dropping",
item->mlineindex);
}
} else {
IceCandidateItem new;
new.mlineindex = item->mlineindex;
new.candidate = g_strdup (item->candidate);
new.promise = NULL;
GST_INFO_OBJECT (webrtc, "Unknown mline %u, deferring", item->mlineindex);
ICE_LOCK (webrtc);
g_array_append_val (webrtc->priv->pending_remote_ice_candidates, new);
ICE_UNLOCK (webrtc);
}
return;
}
GST_LOG_OBJECT (webrtc, "adding ICE candidate with mline:%u, %s",
item->mlineindex, item->candidate);
gst_webrtc_ice_add_candidate (webrtc->priv->ice, stream, item->candidate,
item->promise);
}
static void
_add_ice_candidates_from_sdp (GstWebRTCBin * webrtc, gint mlineindex,
const GstSDPMedia * media)
{
gint a;
GstWebRTCICEStream *stream = NULL;
for (a = 0; a < gst_sdp_media_attributes_len (media); a++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, a);
if (g_strcmp0 (attr->key, "candidate") == 0) {
gchar *candidate;
if (stream == NULL)
stream = _find_ice_stream_for_session (webrtc, mlineindex);
if (stream == NULL) {
GST_DEBUG_OBJECT (webrtc,
"Unknown mline %u, dropping ICE candidates from SDP", mlineindex);
return;
}
candidate = g_strdup_printf ("a=candidate:%s", attr->value);
GST_LOG_OBJECT (webrtc, "adding ICE candidate with mline:%u, %s",
mlineindex, candidate);
gst_webrtc_ice_add_candidate (webrtc->priv->ice, stream, candidate, NULL);
g_free (candidate);
}
}
}
static void
_add_ice_candidate_to_sdp (GstWebRTCBin * webrtc,
GstSDPMessage * sdp, gint mline_index, const gchar * candidate)
{
GstSDPMedia *media = NULL;
if (mline_index < sdp->medias->len) {
media = &g_array_index (sdp->medias, GstSDPMedia, mline_index);
}
if (media == NULL) {
GST_WARNING_OBJECT (webrtc, "Couldn't find mline %d to merge ICE candidate",
mline_index);
return;
}
// Add the candidate as an attribute, first stripping off the existing
// candidate: key from the string description
if (strlen (candidate) < 10) {
GST_WARNING_OBJECT (webrtc,
"Dropping invalid ICE candidate for mline %d: %s", mline_index,
candidate);
return;
}
gst_sdp_media_add_attribute (media, "candidate", candidate + 10);
}
static void
_add_end_of_candidate_to_sdp (GstWebRTCBin * webrtc,
GstSDPMessage * sdp, gint mline_index)
{
GstSDPMedia *media = NULL;
if (mline_index < sdp->medias->len) {
media = &g_array_index (sdp->medias, GstSDPMedia, mline_index);
}
if (media == NULL) {
GST_WARNING_OBJECT (webrtc, "Couldn't find mline %d to merge ICE candidate",
mline_index);
return;
}
gst_sdp_media_add_attribute (media, "end-of-candidates", "");
}
static gboolean
_filter_sdp_fields (GQuark field_id, const GValue * value,
GstStructure * new_structure)
{
if (!g_str_has_prefix (g_quark_to_string (field_id), "a-")) {
gst_structure_id_set_value (new_structure, field_id, value);
}
return TRUE;
}
static guint
transport_stream_ptmap_get_rtp_header_extension_id (TransportStream * stream,
const char *rtphdrext_uri)
{
guint i;
for (i = 0; i < stream->ptmap->len; i++) {
PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
guint id;
id = caps_get_rtp_header_extension_id (item->caps, rtphdrext_uri);
if (id != -1)
return id;
}
return -1;
}
static void
ensure_rtx_hdr_ext (TransportStream * stream)
{
stream->rtphdrext_id_stream_id =
transport_stream_ptmap_get_rtp_header_extension_id (stream,
RTPHDREXT_STREAM_ID);
stream->rtphdrext_id_repaired_stream_id =
transport_stream_ptmap_get_rtp_header_extension_id (stream,
RTPHDREXT_REPAIRED_STREAM_ID);
/* TODO: removing header extensions usage from rtx on renegotiation */
if (stream->rtxsend) {
if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxsend_stream_id) {
stream->rtxsend_stream_id =
gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID);
if (!stream->rtxsend_stream_id)
g_warn_if_reached ();
gst_rtp_header_extension_set_id (stream->rtxsend_stream_id,
stream->rtphdrext_id_stream_id);
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id,
stream->rtphdrext_id_stream_id, stream->rtxsend);
g_signal_emit_by_name (stream->rtxsend, "add-extension",
stream->rtxsend_stream_id);
}
if (stream->rtphdrext_id_repaired_stream_id != -1
&& !stream->rtxsend_repaired_stream_id) {
stream->rtxsend_repaired_stream_id =
gst_rtp_header_extension_create_from_uri
(RTPHDREXT_REPAIRED_STREAM_ID);
if (!stream->rtxsend_repaired_stream_id)
g_warn_if_reached ();
gst_rtp_header_extension_set_id (stream->rtxsend_repaired_stream_id,
stream->rtphdrext_id_repaired_stream_id);
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id,
stream->rtphdrext_id_repaired_stream_id, stream->rtxsend);
g_signal_emit_by_name (stream->rtxsend, "add-extension",
stream->rtxsend_repaired_stream_id);
}
}
if (stream->rtxreceive) {
if (stream->rtphdrext_id_stream_id != -1 && !stream->rtxreceive_stream_id) {
stream->rtxreceive_stream_id =
gst_rtp_header_extension_create_from_uri (RTPHDREXT_STREAM_ID);
if (!stream->rtxreceive_stream_id)
g_warn_if_reached ();
gst_rtp_header_extension_set_id (stream->rtxreceive_stream_id,
stream->rtphdrext_id_stream_id);
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_stream_id,
stream->rtphdrext_id_stream_id, stream->rtxreceive);
g_signal_emit_by_name (stream->rtxreceive, "add-extension",
stream->rtxreceive_stream_id);
}
if (stream->rtphdrext_id_repaired_stream_id != -1
&& !stream->rtxreceive_repaired_stream_id) {
stream->rtxreceive_repaired_stream_id =
gst_rtp_header_extension_create_from_uri
(RTPHDREXT_REPAIRED_STREAM_ID);
if (!stream->rtxreceive_repaired_stream_id)
g_warn_if_reached ();
gst_rtp_header_extension_set_id (stream->rtxreceive_repaired_stream_id,
stream->rtphdrext_id_repaired_stream_id);
GST_DEBUG_OBJECT (stream, "adding rtp header extension %" GST_PTR_FORMAT
" with id %u to %" GST_PTR_FORMAT, stream->rtxsend_repaired_stream_id,
stream->rtphdrext_id_repaired_stream_id, stream->rtxreceive);
g_signal_emit_by_name (stream->rtxreceive, "add-extension",
stream->rtxreceive_repaired_stream_id);
}
}
}
static void
_update_transport_ptmap_from_media (GstWebRTCBin * webrtc,
TransportStream * stream, const GstSDPMessage * sdp, guint media_idx)
{
guint i, len;
const gchar *proto;
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
/* get proto */
proto = gst_sdp_media_get_proto (media);
if (proto != NULL) {
/* Parse global SDP attributes once */
GstCaps *global_caps = gst_caps_new_empty_simple ("application/x-unknown");
GST_DEBUG_OBJECT (webrtc, "mapping sdp session level attributes to caps");
gst_sdp_message_attributes_to_caps (sdp, global_caps);
GST_DEBUG_OBJECT (webrtc, "mapping sdp media level attributes to caps");
gst_sdp_media_attributes_to_caps (media, global_caps);
len = gst_sdp_media_formats_len (media);
for (i = 0; i < len; i++) {
GstCaps *caps, *outcaps;
GstStructure *s;
PtMapItem item;
gint pt;
guint j;
pt = atoi (gst_sdp_media_get_format (media, i));
GST_DEBUG_OBJECT (webrtc, " looking at %d pt: %d", i, pt);
/* convert caps */
caps = gst_sdp_media_get_caps_from_media (media, pt);
if (caps == NULL) {
GST_WARNING_OBJECT (webrtc, " skipping pt %d without caps", pt);
continue;
}
/* Merge in global caps */
/* Intersect will merge in missing fields to the current caps */
outcaps = gst_caps_intersect (caps, global_caps);
gst_caps_unref (caps);
s = gst_caps_get_structure (outcaps, 0);
gst_structure_set_name (s, "application/x-rtp");
if (!g_strcmp0 (gst_structure_get_string (s, "encoding-name"), "ULPFEC"))
gst_structure_set (s, "is-fec", G_TYPE_BOOLEAN, TRUE, NULL);
item.caps = gst_caps_new_empty ();
for (j = 0; j < gst_caps_get_size (outcaps); j++) {
GstStructure *s = gst_caps_get_structure (outcaps, j);
GstStructure *filtered =
gst_structure_new_empty (gst_structure_get_name (s));
gst_structure_foreach (s,
(GstStructureForeachFunc) _filter_sdp_fields, filtered);
gst_caps_append_structure (item.caps, filtered);
}
item.pt = pt;
item.media_idx = media_idx;
gst_caps_unref (outcaps);
g_array_append_val (stream->ptmap, item);
}
gst_caps_unref (global_caps);
}
}
static void
_update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
const GstSDPMessage * sdp, guint media_idx,
TransportStream * stream, GstWebRTCRTPTransceiver * rtp_trans,
GStrv bundled, guint bundle_idx, GError ** error)
{
WebRTCTransceiver *trans = WEBRTC_TRANSCEIVER (rtp_trans);
GstWebRTCRTPTransceiverDirection prev_dir = rtp_trans->current_direction;
GstWebRTCRTPTransceiverDirection new_dir;
const GstSDPMedia *local_media, *remote_media;
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
GstWebRTCDTLSSetup new_setup;
char *local_msid = NULL;
gboolean new_rtcp_rsize;
ReceiveState receive_state = RECEIVE_STATE_UNSET;
int i;
local_media =
gst_sdp_message_get_media (webrtc->current_local_description->sdp,
media_idx);
remote_media =
gst_sdp_message_get_media (webrtc->current_remote_description->sdp,
media_idx);
rtp_trans->mline = media_idx;
if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio")) {
if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
GST_FIXME_OBJECT (webrtc, "Updating video transceiver %" GST_PTR_FORMAT
" to audio, which isn't fully supported.", rtp_trans);
rtp_trans->kind = GST_WEBRTC_KIND_AUDIO;
}
if (!g_strcmp0 (gst_sdp_media_get_media (media), "video")) {
if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
GST_FIXME_OBJECT (webrtc, "Updating audio transceiver %" GST_PTR_FORMAT
" to video, which isn't fully supported.", rtp_trans);
rtp_trans->kind = GST_WEBRTC_KIND_VIDEO;
}
for (i = 0; i < gst_sdp_media_attributes_len (media); i++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, i);
if (g_strcmp0 (attr->key, "mid") == 0) {
g_free (rtp_trans->mid);
rtp_trans->mid = g_strdup (attr->value);
}
}
{
GstWebRTCRTPTransceiverDirection local_dir, remote_dir;
GstWebRTCDTLSSetup local_setup, remote_setup;
local_setup = _get_dtls_setup_from_media (local_media);
remote_setup = _get_dtls_setup_from_media (remote_media);
new_setup = _get_final_setup (local_setup, remote_setup);
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Cannot intersect direction attributes for media %u", media_idx);
return;
}
local_dir = _get_direction_from_media (local_media);
remote_dir = _get_direction_from_media (remote_media);
new_dir = _get_final_direction (local_dir, remote_dir);
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Cannot intersect dtls setup attributes for media %u", media_idx);
return;
}
#if 0
if (prev_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE
&& new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE
&& prev_dir != new_dir) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"transceiver direction changes are not implemented. Media %u",
media_idx);
return;
}
#endif
if (!bundled || bundle_idx == media_idx) {
new_rtcp_rsize = _media_has_attribute_key (local_media, "rtcp-rsize")
&& _media_has_attribute_key (remote_media, "rtcp-rsize");
{
GObject *session;
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-session",
media_idx, &session);
if (session) {
g_object_set (session, "rtcp-reduced-size", new_rtcp_rsize, NULL);
g_object_unref (session);
}
}
}
}
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) {
if (!bundled) {
/* Not a bundled stream means this entire transport is inactive,
* so set the receive state to BLOCK below */
stream->active = FALSE;
receive_state = RECEIVE_STATE_BLOCK;
}
} else {
/* If this transceiver is active for sending or receiving,
* we still need receive at least RTCP, so need to unblock
* the receive bin below. */
GST_LOG_OBJECT (webrtc, "marking stream %p as active", stream);
receive_state = RECEIVE_STATE_PASS;
stream->active = TRUE;
}
if (new_dir != prev_dir) {
guint rtp_session_id = bundled ? bundle_idx : media_idx;
GST_DEBUG_OBJECT (webrtc, "transceiver %" GST_PTR_FORMAT
" direction change from %s to %s", rtp_trans,
gst_webrtc_rtp_transceiver_direction_to_string (prev_dir),
gst_webrtc_rtp_transceiver_direction_to_string (new_dir));
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) {
GstWebRTCBinPad *pad;
pad = _find_pad_for_mline (webrtc, GST_PAD_SRC, media_idx);
if (pad) {
GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
if (target) {
GstPad *peer = gst_pad_get_peer (target);
if (peer) {
gst_pad_send_event (peer, gst_event_new_eos ());
gst_object_unref (peer);
}
gst_object_unref (target);
}
gst_object_unref (pad);
}
/* XXX: send eos event up the sink pad as well? */
}
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY ||
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GstWebRTCBinPad *pad =
_find_pad_for_transceiver (webrtc, GST_PAD_SINK, rtp_trans);
local_msid = _get_msid_from_media (local_media);
if (pad) {
GST_DEBUG_OBJECT (webrtc, "found existing send pad %" GST_PTR_FORMAT
" for transceiver %" GST_PTR_FORMAT " with msid \'%s\'", pad, trans,
pad->msid);
if (g_strcmp0 (pad->msid, local_msid) != 0) {
GST_DEBUG_OBJECT (webrtc, "send pad %" GST_PTR_FORMAT
" transceiver %" GST_PTR_FORMAT " changing msid from \'%s\'"
" to \'%s\'", pad, trans, pad->msid, local_msid);
g_clear_pointer (&pad->msid, g_free);
pad->msid = local_msid;
g_object_notify (G_OBJECT (pad), "msid");
local_msid = NULL;
} else {
g_clear_pointer (&local_msid, g_free);
}
gst_object_unref (pad);
} else {
GST_DEBUG_OBJECT (webrtc,
"creating new send pad for transceiver %" GST_PTR_FORMAT, trans);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, rtp_trans,
G_MAXUINT, local_msid);
local_msid = NULL;
_connect_input_stream (webrtc, pad);
_add_pad (webrtc, pad);
}
}
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY ||
new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GstWebRTCBinPad *pad =
_find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
char *remote_msid = _get_msid_from_media (remote_media);
if (pad) {
GST_DEBUG_OBJECT (webrtc, "found existing receive pad %" GST_PTR_FORMAT
" for transceiver %" GST_PTR_FORMAT " with msid \'%s\'", pad, trans,
pad->msid);
if (g_strcmp0 (pad->msid, remote_msid) != 0) {
GST_DEBUG_OBJECT (webrtc, "receive pad %" GST_PTR_FORMAT
" transceiver %" GST_PTR_FORMAT " changing msid from \'%s\'"
" to \'%s\'", pad, trans, pad->msid, remote_msid);
g_clear_pointer (&pad->msid, g_free);
pad->msid = remote_msid;
remote_msid = NULL;
g_object_notify (G_OBJECT (pad), "msid");
} else {
g_clear_pointer (&remote_msid, g_free);
}
gst_object_unref (pad);
} else {
GST_DEBUG_OBJECT (webrtc,
"creating new receive pad for transceiver %" GST_PTR_FORMAT, trans);
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SRC, rtp_trans,
G_MAXUINT, remote_msid);
remote_msid = NULL;
if (!trans->stream) {
TransportStream *item;
item =
_get_or_create_transport_stream (webrtc, rtp_session_id, FALSE);
webrtc_transceiver_set_transport (trans, item);
}
_connect_output_stream (webrtc, trans->stream, rtp_session_id);
/* delay adding the pad until rtpbin creates the recv output pad
* to ghost to so queries/events travel through the pipeline correctly
* as soon as the pad is added */
_add_pad_to_list (webrtc, pad);
}
}
rtp_trans->mline = media_idx;
rtp_trans->current_direction = new_dir;
}
if (!bundled || bundle_idx == media_idx) {
if (stream->rtxsend || stream->rtxreceive) {
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
}
g_object_set (stream, "dtls-client",
new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL);
}
/* Must be after setting the "dtls-client" so that data is not pushed into
* the dtlssrtp elements before the ssl direction has been set which will
* throw SSL errors */
if (receive_state != RECEIVE_STATE_UNSET)
transport_receive_bin_set_receive_state (stream->receive_bin,
receive_state);
}
/* must be called with the pc lock held */
static gint
_generate_data_channel_id (GstWebRTCBin * webrtc)
{
gboolean is_client;
gint new_id = -1, max_channels = 0;
if (webrtc->priv->sctp_transport) {
g_object_get (webrtc->priv->sctp_transport, "max-channels", &max_channels,
NULL);
}
if (max_channels <= 0) {
max_channels = 65534;
}
g_object_get (webrtc->priv->sctp_transport->transport, "client", &is_client,
NULL);
/* TODO: a better search algorithm */
do {
WebRTCDataChannel *channel;
new_id++;
if (new_id < 0 || new_id >= max_channels) {
/* exhausted id space */
GST_WARNING_OBJECT (webrtc, "Could not find a suitable "
"data channel id (max %i)", max_channels);
return -1;
}
/* client must generate even ids, server must generate odd ids */
if (new_id % 2 == !(!is_client))
continue;
channel = _find_data_channel_for_id (webrtc, new_id);
if (!channel)
break;
} while (TRUE);
return new_id;
}
static void
_update_data_channel_from_sdp_media (GstWebRTCBin * webrtc,
const GstSDPMessage * sdp, guint media_idx, TransportStream * stream,
GError ** error)
{
const GstSDPMedia *local_media, *remote_media;
GstWebRTCDTLSSetup local_setup, remote_setup, new_setup;
TransportReceiveBin *receive;
int local_port, remote_port;
guint64 local_max_size, remote_max_size, max_size;
int i;
local_media =
gst_sdp_message_get_media (webrtc->current_local_description->sdp,
media_idx);
remote_media =
gst_sdp_message_get_media (webrtc->current_remote_description->sdp,
media_idx);
local_setup = _get_dtls_setup_from_media (local_media);
remote_setup = _get_dtls_setup_from_media (remote_media);
new_setup = _get_final_setup (local_setup, remote_setup);
if (new_setup == GST_WEBRTC_DTLS_SETUP_NONE) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Cannot intersect dtls setup for media %u", media_idx);
return;
}
/* data channel is always rtcp-muxed to avoid generating ICE candidates
* for RTCP */
g_object_set (stream, "dtls-client",
new_setup == GST_WEBRTC_DTLS_SETUP_ACTIVE, NULL);
local_port = _get_sctp_port_from_media (local_media);
remote_port = _get_sctp_port_from_media (local_media);
if (local_port == -1 || remote_port == -1) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Could not find sctp port for media %u (local %i, remote %i)",
media_idx, local_port, remote_port);
return;
}
if (0 == (local_max_size =
_get_sctp_max_message_size_from_media (local_media)))
local_max_size = G_MAXUINT64;
if (0 == (remote_max_size =
_get_sctp_max_message_size_from_media (remote_media)))
remote_max_size = G_MAXUINT64;
max_size = MIN (local_max_size, remote_max_size);
webrtc->priv->sctp_transport->max_message_size = max_size;
{
guint orig_local_port, orig_remote_port;
/* XXX: sctpassociation warns if we are in the wrong state */
g_object_get (webrtc->priv->sctp_transport->sctpdec, "local-sctp-port",
&orig_local_port, NULL);
if (orig_local_port != local_port)
g_object_set (webrtc->priv->sctp_transport->sctpdec, "local-sctp-port",
local_port, NULL);
g_object_get (webrtc->priv->sctp_transport->sctpenc, "remote-sctp-port",
&orig_remote_port, NULL);
if (orig_remote_port != remote_port)
g_object_set (webrtc->priv->sctp_transport->sctpenc, "remote-sctp-port",
remote_port, NULL);
}
DC_LOCK (webrtc);
for (i = 0; i < webrtc->priv->data_channels->len; i++) {
WebRTCDataChannel *channel;
channel = g_ptr_array_index (webrtc->priv->data_channels, i);
if (channel->parent.id == -1)
channel->parent.id = _generate_data_channel_id (webrtc);
if (channel->parent.id == -1)
GST_ELEMENT_WARNING (webrtc, RESOURCE, NOT_FOUND,
("%s", "Failed to generate an identifier for a data channel"), NULL);
if (webrtc->priv->sctp_transport->association_established
&& !channel->parent.negotiated && !channel->opened) {
webrtc_data_channel_link_to_sctp (channel, webrtc->priv->sctp_transport);
webrtc_data_channel_start_negotiation (channel);
}
}
DC_UNLOCK (webrtc);
stream->active = TRUE;
receive = TRANSPORT_RECEIVE_BIN (stream->receive_bin);
transport_receive_bin_set_receive_state (receive, RECEIVE_STATE_PASS);
}
static gboolean
_find_compatible_unassociated_transceiver (GstWebRTCRTPTransceiver * p1,
gconstpointer data)
{
GstWebRTCKind kind = GPOINTER_TO_INT (data);
if (p1->mid)
return FALSE;
if (p1->mline != -1)
return FALSE;
if (p1->stopped)
return FALSE;
if (p1->kind != GST_WEBRTC_KIND_UNKNOWN && p1->kind != kind)
return FALSE;
return TRUE;
}
static void
_connect_rtpfunnel (GstWebRTCBin * webrtc, guint session_id)
{
gchar *pad_name;
GstPad *srcpad;
GstPad *rtp_sink;
TransportStream *stream = _find_transport_for_session (webrtc, session_id);
g_assert (stream);
if (webrtc->rtpfunnel)
goto done;
webrtc->rtpfunnel = gst_element_factory_make ("rtpfunnel", NULL);
gst_bin_add (GST_BIN (webrtc), webrtc->rtpfunnel);
gst_element_sync_state_with_parent (webrtc->rtpfunnel);
srcpad = gst_element_get_static_pad (webrtc->rtpfunnel, "src");
pad_name = g_strdup_printf ("send_rtp_sink_%d", session_id);
rtp_sink = gst_element_request_pad_simple (webrtc->rtpbin, pad_name);
g_free (pad_name);
gst_pad_link (srcpad, rtp_sink);
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);
done:
return;
}
static gboolean
_update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
GstWebRTCSessionDescription * sdp, GError ** error)
{
int i;
gboolean ret = FALSE;
GStrv bundled = NULL;
guint bundle_idx = 0;
TransportStream *bundle_stream = NULL;
/* FIXME: With some peers, it's possible we could have
* multiple bundles to deal with, although I've never seen one yet */
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE)
if (!_parse_bundle (sdp->sdp, &bundled, error))
goto done;
if (bundled) {
if (!_get_bundle_index (sdp->sdp, bundled, &bundle_idx)) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Bundle tag is %s but no media found matching", bundled[0]);
goto done;
}
bundle_stream = _get_or_create_transport_stream (webrtc, bundle_idx,
_message_media_is_datachannel (sdp->sdp, bundle_idx));
/* Mark the bundle stream as inactive to start. It will be set to TRUE
* by any bundled mline that is active, and at the end we set the
* receivebin to BLOCK if all mlines were inactive. */
bundle_stream->active = FALSE;
g_array_set_size (bundle_stream->ptmap, 0);
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
/* When bundling, we need to do this up front, or else RTX
* parameters aren't set up properly for the bundled streams */
_update_transport_ptmap_from_media (webrtc, bundle_stream, sdp->sdp, i);
}
ensure_rtx_hdr_ext (bundle_stream);
_connect_rtpfunnel (webrtc, bundle_idx);
}
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
TransportStream *stream;
GstWebRTCRTPTransceiver *trans;
guint transport_idx;
/* skip rejected media */
if (gst_sdp_media_get_port (media) == 0)
continue;
if (bundled)
transport_idx = bundle_idx;
else
transport_idx = i;
trans = _find_transceiver_for_sdp_media (webrtc, sdp->sdp, i);
stream = _get_or_create_transport_stream (webrtc, transport_idx,
_message_media_is_datachannel (sdp->sdp, transport_idx));
if (!bundled) {
/* When bundling, these were all set up above, but when not
* bundling we need to do it now */
g_array_set_size (stream->ptmap, 0);
_update_transport_ptmap_from_media (webrtc, stream, sdp->sdp, i);
ensure_rtx_hdr_ext (stream);
}
if (trans)
webrtc_transceiver_set_transport ((WebRTCTransceiver *) trans, stream);
if (source == SDP_LOCAL && sdp->type == GST_WEBRTC_SDP_TYPE_OFFER && !trans) {
g_set_error (error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"State mismatch. Could not find local transceiver by mline %u", i);
goto done;
} else {
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
/* No existing transceiver, find an unused one */
if (!trans) {
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0)
kind = GST_WEBRTC_KIND_AUDIO;
else if (g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0)
kind = GST_WEBRTC_KIND_VIDEO;
else
GST_LOG_OBJECT (webrtc, "Unknown media kind %s",
GST_STR_NULL (gst_sdp_media_get_media (media)));
trans = _find_transceiver (webrtc, GINT_TO_POINTER (kind),
(FindTransceiverFunc) _find_compatible_unassociated_transceiver);
}
/* Still no transceiver? Create one */
/* XXX: default to the advertised direction in the sdp for new
* transceivers. The spec doesn't actually say what happens here, only
* that calls to setDirection will change the value. Nothing about
* a default value when the transceiver is created internally */
if (!trans) {
WebRTCTransceiver *t = _create_webrtc_transceiver (webrtc,
_get_direction_from_media (media), i, kind, NULL);
webrtc_transceiver_set_transport (t, stream);
trans = GST_WEBRTC_RTP_TRANSCEIVER (t);
PC_UNLOCK (webrtc);
g_signal_emit (webrtc,
gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0, trans);
PC_LOCK (webrtc);
}
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
trans, bundled, bundle_idx, error);
if (error && *error)
goto done;
} else if (_message_media_is_datachannel (sdp->sdp, i)) {
_update_data_channel_from_sdp_media (webrtc, sdp->sdp, i, stream,
error);
if (error && *error)
goto done;
} else {
GST_ERROR_OBJECT (webrtc, "Unknown media type in SDP at index %u", i);
}
}
}
if (bundle_stream && bundle_stream->active == FALSE) {
/* No bundled mline marked the bundle as active, so block the receive bin, as
* this bundle is completely inactive */
GST_LOG_OBJECT (webrtc,
"All mlines in bundle %u are inactive. Blocking receiver", bundle_idx);
transport_receive_bin_set_receive_state (bundle_stream->receive_bin,
RECEIVE_STATE_BLOCK);
}
ret = TRUE;
done:
g_strfreev (bundled);
return ret;
}
static gint
transceivers_media_num_cmp (GstWebRTCBin * webrtc,
GstWebRTCSessionDescription * previous, GstWebRTCSessionDescription * new)
{
if (!previous)
return 0;
return gst_sdp_message_medias_len (new->sdp) -
gst_sdp_message_medias_len (previous->sdp);
}
static gboolean
check_locked_mlines (GstWebRTCBin * webrtc, GstWebRTCSessionDescription * sdp,
GError ** error)
{
guint i;
for (i = 0; i < gst_sdp_message_medias_len (sdp->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sdp->sdp, i);
GstWebRTCRTPTransceiver *rtp_trans;
WebRTCTransceiver *trans;
rtp_trans = _find_transceiver_for_sdp_media (webrtc, sdp->sdp, i);
/* only look for matching mid */
if (rtp_trans == NULL)
continue;
trans = WEBRTC_TRANSCEIVER (rtp_trans);
/* We only validate the locked mlines for now */
if (!trans->mline_locked)
continue;
if (rtp_trans->mline != i) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"m-line with mid %s is at position %d, but was locked to %d, "
"rejecting", rtp_trans->mid, i, rtp_trans->mline);
return FALSE;
}
if (rtp_trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
if (!g_strcmp0 (gst_sdp_media_get_media (media), "audio") &&
rtp_trans->kind != GST_WEBRTC_KIND_AUDIO) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"m-line %d with transceiver <%s> was locked to %s, but SDP has "
"%s media", i, GST_OBJECT_NAME (rtp_trans),
gst_webrtc_kind_to_string (rtp_trans->kind),
gst_sdp_media_get_media (media));
return FALSE;
}
if (!g_strcmp0 (gst_sdp_media_get_media (media), "video") &&
rtp_trans->kind != GST_WEBRTC_KIND_VIDEO) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_INTERNAL_FAILURE,
"m-line %d with transceiver <%s> was locked to %s, but SDP has "
"%s media", i, GST_OBJECT_NAME (rtp_trans),
gst_webrtc_kind_to_string (rtp_trans->kind),
gst_sdp_media_get_media (media));
return FALSE;
}
}
}
return TRUE;
}
struct set_description
{
SDPSource source;
GstWebRTCSessionDescription *sdp;
};
static GstWebRTCSessionDescription *
get_previous_description (GstWebRTCBin * webrtc, SDPSource source,
GstWebRTCSDPType type)
{
switch (type) {
case GST_WEBRTC_SDP_TYPE_OFFER:
case GST_WEBRTC_SDP_TYPE_PRANSWER:
case GST_WEBRTC_SDP_TYPE_ANSWER:
if (source == SDP_LOCAL) {
return webrtc->current_local_description;
} else {
return webrtc->current_remote_description;
}
case GST_WEBRTC_SDP_TYPE_ROLLBACK:
return NULL;
default:
/* other values mean memory corruption/uninitialized! */
g_assert_not_reached ();
break;
}
return NULL;
}
static GstWebRTCSessionDescription *
get_last_generated_description (GstWebRTCBin * webrtc, SDPSource source,
GstWebRTCSDPType type)
{
switch (type) {
case GST_WEBRTC_SDP_TYPE_OFFER:
if (source == SDP_REMOTE)
return webrtc->priv->last_generated_answer;
else
return webrtc->priv->last_generated_offer;
break;
case GST_WEBRTC_SDP_TYPE_PRANSWER:
case GST_WEBRTC_SDP_TYPE_ANSWER:
if (source == SDP_LOCAL)
return webrtc->priv->last_generated_answer;
else
return webrtc->priv->last_generated_offer;
case GST_WEBRTC_SDP_TYPE_ROLLBACK:
return NULL;
default:
/* other values mean memory corruption/uninitialized! */
g_assert_not_reached ();
break;
}
return NULL;
}
/* http://w3c.github.io/webrtc-pc/#set-description */
static GstStructure *
_set_description_task (GstWebRTCBin * webrtc, struct set_description *sd)
{
GstWebRTCSignalingState old_signaling_state = webrtc->signaling_state;
GstWebRTCSignalingState new_signaling_state = webrtc->signaling_state;
gboolean signalling_state_changed = FALSE;
GError *error = NULL;
GStrv bundled = NULL;
guint bundle_idx = 0;
guint i;
{
const gchar *state = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE,
webrtc->signaling_state);
const gchar *type_str =
_enum_value_to_string (GST_TYPE_WEBRTC_SDP_TYPE, sd->sdp->type);
gchar *sdp_text = gst_sdp_message_as_text (sd->sdp->sdp);
GST_INFO_OBJECT (webrtc, "Attempting to set %s %s in the %s state",
_sdp_source_to_string (sd->source), type_str, state);
GST_TRACE_OBJECT (webrtc, "SDP contents\n%s", sdp_text);
g_free (sdp_text);
}
if (!validate_sdp (webrtc->signaling_state, sd->source, sd->sdp, &error))
goto out;
if (webrtc->bundle_policy != GST_WEBRTC_BUNDLE_POLICY_NONE)
if (!_parse_bundle (sd->sdp->sdp, &bundled, &error))
goto out;
if (bundled) {
if (!_get_bundle_index (sd->sdp->sdp, bundled, &bundle_idx)) {
g_set_error (&error, GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Bundle tag is %s but no matching media found", bundled[0]);
goto out;
}
}
if (transceivers_media_num_cmp (webrtc,
get_previous_description (webrtc, sd->source, sd->sdp->type),
sd->sdp) < 0) {
g_set_error_literal (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"m=lines removed from the SDP. Processing a completely new connection "
"is not currently supported.");
goto out;
}
if ((sd->sdp->type == GST_WEBRTC_SDP_TYPE_PRANSWER ||
sd->sdp->type == GST_WEBRTC_SDP_TYPE_ANSWER) &&
transceivers_media_num_cmp (webrtc,
get_last_generated_description (webrtc, sd->source, sd->sdp->type),
sd->sdp) != 0) {
g_set_error_literal (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_SDP_SYNTAX_ERROR,
"Answer doesn't have the same number of m-lines as the offer.");
goto out;
}
if (!check_locked_mlines (webrtc, sd->sdp, &error))
goto out;
switch (sd->sdp->type) {
case GST_WEBRTC_SDP_TYPE_OFFER:{
if (sd->source == SDP_LOCAL) {
if (webrtc->pending_local_description)
gst_webrtc_session_description_free
(webrtc->pending_local_description);
webrtc->pending_local_description =
gst_webrtc_session_description_copy (sd->sdp);
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER;
} else {
if (webrtc->pending_remote_description)
gst_webrtc_session_description_free
(webrtc->pending_remote_description);
webrtc->pending_remote_description =
gst_webrtc_session_description_copy (sd->sdp);
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER;
}
break;
}
case GST_WEBRTC_SDP_TYPE_ANSWER:{
if (sd->source == SDP_LOCAL) {
if (webrtc->current_local_description)
gst_webrtc_session_description_free
(webrtc->current_local_description);
webrtc->current_local_description =
gst_webrtc_session_description_copy (sd->sdp);
if (webrtc->current_remote_description)
gst_webrtc_session_description_free
(webrtc->current_remote_description);
webrtc->current_remote_description = webrtc->pending_remote_description;
webrtc->pending_remote_description = NULL;
} else {
if (webrtc->current_remote_description)
gst_webrtc_session_description_free
(webrtc->current_remote_description);
webrtc->current_remote_description =
gst_webrtc_session_description_copy (sd->sdp);
if (webrtc->current_local_description)
gst_webrtc_session_description_free
(webrtc->current_local_description);
webrtc->current_local_description = webrtc->pending_local_description;
webrtc->pending_local_description = NULL;
}
if (webrtc->pending_local_description)
gst_webrtc_session_description_free (webrtc->pending_local_description);
webrtc->pending_local_description = NULL;
if (webrtc->pending_remote_description)
gst_webrtc_session_description_free
(webrtc->pending_remote_description);
webrtc->pending_remote_description = NULL;
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_STABLE;
break;
}
case GST_WEBRTC_SDP_TYPE_ROLLBACK:{
GST_FIXME_OBJECT (webrtc, "rollbacks are completely untested");
if (sd->source == SDP_LOCAL) {
if (webrtc->pending_local_description)
gst_webrtc_session_description_free
(webrtc->pending_local_description);
webrtc->pending_local_description = NULL;
} else {
if (webrtc->pending_remote_description)
gst_webrtc_session_description_free
(webrtc->pending_remote_description);
webrtc->pending_remote_description = NULL;
}
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_STABLE;
break;
}
case GST_WEBRTC_SDP_TYPE_PRANSWER:{
GST_FIXME_OBJECT (webrtc, "pranswers are completely untested");
if (sd->source == SDP_LOCAL) {
if (webrtc->pending_local_description)
gst_webrtc_session_description_free
(webrtc->pending_local_description);
webrtc->pending_local_description =
gst_webrtc_session_description_copy (sd->sdp);
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER;
} else {
if (webrtc->pending_remote_description)
gst_webrtc_session_description_free
(webrtc->pending_remote_description);
webrtc->pending_remote_description =
gst_webrtc_session_description_copy (sd->sdp);
new_signaling_state = GST_WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER;
}
break;
}
}
if (sd->sdp->type == GST_WEBRTC_SDP_TYPE_ROLLBACK) {
/* FIXME:
* If the mid value of an RTCRtpTransceiver was set to a non-null value
* by the RTCSessionDescription that is being rolled back, set the mid
* value of that transceiver to null, as described by [JSEP]
* (section 4.1.7.2.).
* If an RTCRtpTransceiver was created by applying the
* RTCSessionDescription that is being rolled back, and a track has not
* been attached to it via addTrack, remove that transceiver from
* connection's set of transceivers, as described by [JSEP]
* (section 4.1.7.2.).
* Restore the value of connection's [[ sctpTransport]] internal slot
* to its value at the last stable signaling state.
*/
}
if (webrtc->signaling_state != new_signaling_state) {
webrtc->signaling_state = new_signaling_state;
signalling_state_changed = TRUE;
}
{
gboolean ice_controller = FALSE;
/* get the current value so we don't change ice controller from TRUE to
* FALSE on renegotiation or once set to TRUE for the initial local offer */
ice_controller = gst_webrtc_ice_get_is_controller (webrtc->priv->ice);
/* we control ice negotiation if we send the initial offer */
ice_controller |=
new_signaling_state == GST_WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER
&& webrtc->current_remote_description == NULL;
/* or, if the remote is an ice-lite peer */
ice_controller |= new_signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE
&& webrtc->current_remote_description
&& _message_has_attribute_key (webrtc->current_remote_description->sdp,
"ice-lite");
GST_DEBUG_OBJECT (webrtc, "we are in ice controlling mode: %s",
ice_controller ? "true" : "false");
gst_webrtc_ice_set_is_controller (webrtc->priv->ice, ice_controller);
}
if (new_signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE) {
GList *tmp;
/* media modifications */
if (!_update_transceivers_from_sdp (webrtc, sd->source, sd->sdp, &error))
goto out;
for (tmp = webrtc->priv->pending_sink_transceivers; tmp;) {
GstWebRTCBinPad *pad = GST_WEBRTC_BIN_PAD (tmp->data);
GstWebRTCRTPTransceiverDirection new_dir;
GList *old = tmp;
const GstSDPMedia *media;
if (!pad->received_caps) {
GST_LOG_OBJECT (pad, "has not received any caps yet. Skipping.");
tmp = tmp->next;
continue;
}
if (!pad->trans) {
GST_LOG_OBJECT (pad, "doesn't have a transceiver");
tmp = tmp->next;
continue;
}
if (pad->trans->mline >= gst_sdp_message_medias_len (sd->sdp->sdp)) {
GST_DEBUG_OBJECT (pad, "not mentioned in this description. Skipping");
tmp = tmp->next;
continue;
}
media = gst_sdp_message_get_media (sd->sdp->sdp, pad->trans->mline);
/* skip rejected media */
if (gst_sdp_media_get_port (media) == 0) {
/* FIXME: arrange for an appropriate flow return */
GST_FIXME_OBJECT (pad, "Media has been rejected. Need to arrange for "
"a more correct flow return.");
tmp = tmp->next;
continue;
}
new_dir = pad->trans->direction;
if (new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY &&
new_dir != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV) {
GST_LOG_OBJECT (pad, "transceiver %" GST_PTR_FORMAT " is not sending "
"data at the moment. Not connecting input stream yet", pad->trans);
tmp = tmp->next;
continue;
}
GST_LOG_OBJECT (pad, "Connecting input stream to rtpbin with "
"transceiver %" GST_PTR_FORMAT " and caps %" GST_PTR_FORMAT,
pad->trans, pad->received_caps);
_connect_input_stream (webrtc, pad);
gst_pad_remove_probe (GST_PAD (pad), pad->block_id);
pad->block_id = 0;
tmp = tmp->next;
gst_object_unref (old->data);
webrtc->priv->pending_sink_transceivers =
g_list_delete_link (webrtc->priv->pending_sink_transceivers, old);
}
}
for (i = 0; i < gst_sdp_message_medias_len (sd->sdp->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (sd->sdp->sdp, i);
gchar *ufrag, *pwd;
TransportStream *item;
guint rtp_session_id = bundled ? bundle_idx : i;
item =
_get_or_create_transport_stream (webrtc, rtp_session_id,
_message_media_is_datachannel (sd->sdp->sdp, rtp_session_id));
if (sd->source == SDP_REMOTE) {
guint j;
for (j = 0; j < gst_sdp_media_attributes_len (media); j++) {
const GstSDPAttribute *attr = gst_sdp_media_get_attribute (media, j);
if (g_strcmp0 (attr->key, "ssrc") == 0) {
GStrv split = g_strsplit (attr->value, " ", 0);
guint32 ssrc;
if (split[0] && sscanf (split[0], "%u", &ssrc) && split[1]
&& g_str_has_prefix (split[1], "cname:")) {
if (!find_mid_ssrc_for_ssrc (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY,
rtp_session_id, ssrc))
transport_stream_add_ssrc_map_item (item,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, ssrc, i);
}
g_strfreev (split);
}
}
}
if (sd->source == SDP_LOCAL && (!bundled || bundle_idx == i)) {
_get_ice_credentials_from_sdp_media (sd->sdp->sdp, i, &ufrag, &pwd);
gst_webrtc_ice_set_local_credentials (webrtc->priv->ice,
item->stream, ufrag, pwd);
g_free (ufrag);
g_free (pwd);
} else if (sd->source == SDP_REMOTE && !_media_is_bundle_only (media)) {
_get_ice_credentials_from_sdp_media (sd->sdp->sdp, i, &ufrag, &pwd);
gst_webrtc_ice_set_remote_credentials (webrtc->priv->ice,
item->stream, ufrag, pwd);
g_free (ufrag);
g_free (pwd);
}
}
if (sd->source == SDP_LOCAL) {
for (i = 0; i < webrtc->priv->ice_stream_map->len; i++) {
IceStreamItem *item =
&g_array_index (webrtc->priv->ice_stream_map, IceStreamItem, i);
gst_webrtc_ice_gather_candidates (webrtc->priv->ice, item->stream);
}
}
/* Add any pending trickle ICE candidates if we have both offer and answer */
if (webrtc->current_local_description && webrtc->current_remote_description) {
int i;
GstWebRTCSessionDescription *remote_sdp =
webrtc->current_remote_description;
/* Add any remote ICE candidates from the remote description to
* support non-trickle peers first */
for (i = 0; i < gst_sdp_message_medias_len (remote_sdp->sdp); i++) {
const GstSDPMedia *media = gst_sdp_message_get_media (remote_sdp->sdp, i);
_add_ice_candidates_from_sdp (webrtc, i, media);
}
ICE_LOCK (webrtc);
for (i = 0; i < webrtc->priv->pending_remote_ice_candidates->len; i++) {
IceCandidateItem *item =
&g_array_index (webrtc->priv->pending_remote_ice_candidates,
IceCandidateItem, i);
_add_ice_candidate (webrtc, item, TRUE);
}
g_array_set_size (webrtc->priv->pending_remote_ice_candidates, 0);
ICE_UNLOCK (webrtc);
}
/*
* If connection's signaling state changed above, fire an event named
* signalingstatechange at connection.
*/
if (signalling_state_changed) {
const gchar *from = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE,
old_signaling_state);
const gchar *to = _enum_value_to_string (GST_TYPE_WEBRTC_SIGNALING_STATE,
new_signaling_state);
GST_TRACE_OBJECT (webrtc, "notify signaling-state from %s "
"to %s", from, to);
PC_UNLOCK (webrtc);
g_object_notify (G_OBJECT (webrtc), "signaling-state");
PC_LOCK (webrtc);
}
if (webrtc->signaling_state == GST_WEBRTC_SIGNALING_STATE_STABLE) {
gboolean prev_need_negotiation = webrtc->priv->need_negotiation;
/* If connection's signaling state is now stable, update the
* negotiation-needed flag. If connection's [[ needNegotiation]] slot
* was true both before and after this update, queue a task to check
* connection's [[needNegotiation]] slot and, if still true, fire a
* simple event named negotiationneeded at connection.*/
_update_need_negotiation (webrtc);
if (prev_need_negotiation && webrtc->priv->need_negotiation) {
_check_need_negotiation_task (webrtc, NULL);
}
}
out:
g_strfreev (bundled);
if (error) {
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
GST_WARNING_OBJECT (webrtc, "returning error: %s", error->message);
g_clear_error (&error);
return s;
} else {
return NULL;
}
}
static void
_free_set_description_data (struct set_description *sd)
{
if (sd->sdp)
gst_webrtc_session_description_free (sd->sdp);
g_free (sd);
}
static void
gst_webrtc_bin_set_remote_description (GstWebRTCBin * webrtc,
GstWebRTCSessionDescription * remote_sdp, GstPromise * promise)
{
struct set_description *sd;
if (remote_sdp == NULL)
goto bad_input;
if (remote_sdp->sdp == NULL)
goto bad_input;
sd = g_new0 (struct set_description, 1);
sd->source = SDP_REMOTE;
sd->sdp = gst_webrtc_session_description_copy (remote_sdp);
if (!gst_webrtc_bin_enqueue_task (webrtc,
(GstWebRTCBinFunc) _set_description_task, sd,
(GDestroyNotify) _free_set_description_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not set remote description. webrtcbin is closed.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
gst_promise_reply (promise, s);
g_clear_error (&error);
}
return;
bad_input:
{
gst_promise_reply (promise, NULL);
g_return_if_reached ();
}
}
static void
gst_webrtc_bin_set_local_description (GstWebRTCBin * webrtc,
GstWebRTCSessionDescription * local_sdp, GstPromise * promise)
{
struct set_description *sd;
if (local_sdp == NULL)
goto bad_input;
if (local_sdp->sdp == NULL)
goto bad_input;
sd = g_new0 (struct set_description, 1);
sd->source = SDP_LOCAL;
sd->sdp = gst_webrtc_session_description_copy (local_sdp);
if (!gst_webrtc_bin_enqueue_task (webrtc,
(GstWebRTCBinFunc) _set_description_task, sd,
(GDestroyNotify) _free_set_description_data, promise)) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not set local description. webrtcbin is closed");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
gst_promise_reply (promise, s);
g_clear_error (&error);
}
return;
bad_input:
{
gst_promise_reply (promise, NULL);
g_return_if_reached ();
}
}
static GstStructure *
_add_ice_candidate_task (GstWebRTCBin * webrtc, IceCandidateItem * item)
{
if (!webrtc->current_local_description || !webrtc->current_remote_description) {
IceCandidateItem new;
new.mlineindex = item->mlineindex;
new.candidate = g_steal_pointer (&item->candidate);
new.promise = NULL;
ICE_LOCK (webrtc);
g_array_append_val (webrtc->priv->pending_remote_ice_candidates, new);
ICE_UNLOCK (webrtc);
} else {
_add_ice_candidate (webrtc, item, FALSE);
}
return NULL;
}
static void
_free_ice_candidate_item (IceCandidateItem * item)
{
_clear_ice_candidate_item (item);
g_free (item);
}
static void
gst_webrtc_bin_add_ice_candidate (GstWebRTCBin * webrtc, guint mline,
const gchar * attr, GstPromise * promise)
{
IceCandidateItem *item;
item = g_new0 (IceCandidateItem, 1);
item->mlineindex = mline;
item->promise = promise ? gst_promise_ref (promise) : NULL;
if (attr && attr[0] != 0) {
if (!g_ascii_strncasecmp (attr, "a=candidate:", 12))
item->candidate = g_strdup (attr);
else if (!g_ascii_strncasecmp (attr, "candidate:", 10))
item->candidate = g_strdup_printf ("a=%s", attr);
}
if (!gst_webrtc_bin_enqueue_task (webrtc,
(GstWebRTCBinFunc) _add_ice_candidate_task, item,
(GDestroyNotify) _free_ice_candidate_item, promise)) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not add ICE candidate. webrtcbin is closed");
GstStructure *s = gst_structure_new ("application/x-gst-promise", "error",
G_TYPE_ERROR, error, NULL);
gst_promise_reply (promise, s);
g_clear_error (&error);
}
}
static GstStructure *
_on_local_ice_candidate_task (GstWebRTCBin * webrtc)
{
gsize i;
GArray *items;
ICE_LOCK (webrtc);
if (webrtc->priv->pending_local_ice_candidates->len == 0) {
ICE_UNLOCK (webrtc);
GST_LOG_OBJECT (webrtc, "No ICE candidates to process right now");
return NULL; /* Nothing to process */
}
/* Take the array so we can process it all and free it later
* without holding the lock
* FIXME: When we depend on GLib 2.64, we can use g_array_steal()
* here */
items = webrtc->priv->pending_local_ice_candidates;
/* Replace with a new array */
webrtc->priv->pending_local_ice_candidates =
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem));
g_array_set_clear_func (webrtc->priv->pending_local_ice_candidates,
(GDestroyNotify) _clear_ice_candidate_item);
ICE_UNLOCK (webrtc);
for (i = 0; i < items->len; i++) {
IceCandidateItem *item = &g_array_index (items, IceCandidateItem, i);
const gchar *cand = item->candidate;
if (cand && !g_ascii_strncasecmp (cand, "a=candidate:", 12)) {
/* stripping away "a=" */
cand += 2;
}
GST_TRACE_OBJECT (webrtc, "produced ICE candidate for mline:%u and %s",
item->mlineindex, cand);
/* First, merge this ice candidate into the appropriate mline
* in the local-description SDP.
* Second, emit the on-ice-candidate signal for the app.
*
* FIXME: This ICE candidate should be stored somewhere with
* the associated mid and also merged back into any subsequent
* local descriptions on renegotiation */
if (webrtc->current_local_description) {
if (cand && cand[0] != '\0') {
_add_ice_candidate_to_sdp (webrtc,
webrtc->current_local_description->sdp, item->mlineindex, cand);
} else {
_add_end_of_candidate_to_sdp (webrtc,
webrtc->current_local_description->sdp, item->mlineindex);
}
}
if (webrtc->pending_local_description) {
if (cand && cand[0] != '\0') {
_add_ice_candidate_to_sdp (webrtc,
webrtc->pending_local_description->sdp, item->mlineindex, cand);
} else {
_add_end_of_candidate_to_sdp (webrtc,
webrtc->pending_local_description->sdp, item->mlineindex);
}
}
PC_UNLOCK (webrtc);
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_ICE_CANDIDATE_SIGNAL],
0, item->mlineindex, cand);
PC_LOCK (webrtc);
}
g_array_free (items, TRUE);
return NULL;
}
static void
_on_local_ice_candidate_cb (GstWebRTCICE * ice, guint session_id,
gchar * candidate, GstWebRTCBin * webrtc)
{
IceCandidateItem item;
gboolean queue_task = FALSE;
item.mlineindex = session_id;
item.candidate = g_strdup (candidate);
item.promise = NULL;
ICE_LOCK (webrtc);
g_array_append_val (webrtc->priv->pending_local_ice_candidates, item);
/* Let the first pending candidate queue a task each time, which will
* handle any that arrive between now and when the task runs */
if (webrtc->priv->pending_local_ice_candidates->len == 1)
queue_task = TRUE;
ICE_UNLOCK (webrtc);
if (queue_task) {
GST_TRACE_OBJECT (webrtc, "Queueing on_ice_candidate_task");
gst_webrtc_bin_enqueue_task (webrtc,
(GstWebRTCBinFunc) _on_local_ice_candidate_task, NULL, NULL, NULL);
}
}
struct get_stats
{
GstPad *pad;
GstPromise *promise;
};
static void
_free_get_stats (struct get_stats *stats)
{
if (stats->pad)
gst_object_unref (stats->pad);
if (stats->promise)
gst_promise_unref (stats->promise);
g_free (stats);
}
/* https://www.w3.org/TR/webrtc/#dom-rtcpeerconnection-getstats() */
static GstStructure *
_get_stats_task (GstWebRTCBin * webrtc, struct get_stats *stats)
{
/* Our selector is the pad,
* https://www.w3.org/TR/webrtc/#dfn-stats-selection-algorithm
*/
return gst_webrtc_bin_create_stats (webrtc, stats->pad);
}
static void
gst_webrtc_bin_get_stats (GstWebRTCBin * webrtc, GstPad * pad,
GstPromise * promise)
{
struct get_stats *stats;
g_return_if_fail (promise != NULL);
g_return_if_fail (pad == NULL || GST_IS_WEBRTC_BIN_PAD (pad));
stats = g_new0 (struct get_stats, 1);
stats->promise = gst_promise_ref (promise);
/* FIXME: check that pad exists in element */
if (pad)
stats->pad = gst_object_ref (pad);
if (!gst_webrtc_bin_enqueue_task (webrtc, (GstWebRTCBinFunc) _get_stats_task,
stats, (GDestroyNotify) _free_get_stats, promise)) {
GError *error =
g_error_new (GST_WEBRTC_ERROR, GST_WEBRTC_ERROR_INVALID_STATE,
"Could not retrieve statistics. webrtcbin is closed.");
GstStructure *s = gst_structure_new ("application/x-gst-promise",
"error", G_TYPE_ERROR, error, NULL);
gst_promise_reply (promise, s);
g_clear_error (&error);
}
}
static GstWebRTCRTPTransceiver *
gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiverDirection direction, GstCaps * caps)
{
WebRTCTransceiver *trans;
g_return_val_if_fail (direction != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
NULL);
PC_LOCK (webrtc);
trans =
_create_webrtc_transceiver (webrtc, direction, -1,
webrtc_kind_from_caps (caps), caps);
GST_LOG_OBJECT (webrtc,
"Created new unassociated transceiver %" GST_PTR_FORMAT, trans);
PC_UNLOCK (webrtc);
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL], 0,
trans);
return gst_object_ref (trans);
}
static void
_deref_and_unref (GstObject ** object)
{
gst_clear_object (object);
}
static GArray *
gst_webrtc_bin_get_transceivers (GstWebRTCBin * webrtc)
{
GArray *arr = g_array_new (FALSE, TRUE, sizeof (GstWebRTCRTPTransceiver *));
int i;
PC_LOCK (webrtc);
g_array_set_clear_func (arr, (GDestroyNotify) _deref_and_unref);
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *trans =
g_ptr_array_index (webrtc->priv->transceivers, i);
gst_object_ref (trans);
g_array_append_val (arr, trans);
}
PC_UNLOCK (webrtc);
return arr;
}
static GstWebRTCRTPTransceiver *
gst_webrtc_bin_get_transceiver (GstWebRTCBin * webrtc, guint idx)
{
GstWebRTCRTPTransceiver *trans = NULL;
PC_LOCK (webrtc);
if (idx >= webrtc->priv->transceivers->len) {
GST_ERROR_OBJECT (webrtc, "No transceiver for idx %d", idx);
goto done;
}
trans = g_ptr_array_index (webrtc->priv->transceivers, idx);
gst_object_ref (trans);
done:
PC_UNLOCK (webrtc);
return trans;
}
static gboolean
gst_webrtc_bin_add_turn_server (GstWebRTCBin * webrtc, const gchar * uri)
{
gboolean ret;
g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), FALSE);
g_return_val_if_fail (uri != NULL, FALSE);
GST_DEBUG_OBJECT (webrtc, "Adding turn server: %s", uri);
PC_LOCK (webrtc);
ret = gst_webrtc_ice_add_turn_server (webrtc->priv->ice, uri);
PC_UNLOCK (webrtc);
return ret;
}
static gboolean
copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
{
GstPad *gpad = GST_PAD_CAST (user_data);
GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
gst_pad_store_sticky_event (gpad, *event);
return TRUE;
}
static WebRTCDataChannel *
gst_webrtc_bin_create_data_channel (GstWebRTCBin * webrtc, const gchar * label,
GstStructure * init_params)
{
gboolean ordered;
gint max_packet_lifetime;
gint max_retransmits;
const gchar *protocol;
gboolean negotiated;
gint id;
GstWebRTCPriorityType priority;
WebRTCDataChannel *ret;
gint max_channels = 65534;
g_return_val_if_fail (GST_IS_WEBRTC_BIN (webrtc), NULL);
g_return_val_if_fail (label != NULL, NULL);
g_return_val_if_fail (strlen (label) <= 65535, NULL);
g_return_val_if_fail (webrtc->priv->is_closed != TRUE, NULL);
if (!init_params
|| !gst_structure_get_boolean (init_params, "ordered", &ordered))
ordered = TRUE;
if (!init_params
|| !gst_structure_get_int (init_params, "max-packet-lifetime",
&max_packet_lifetime))
max_packet_lifetime = -1;
if (!init_params
|| !gst_structure_get_int (init_params, "max-retransmits",
&max_retransmits))
max_retransmits = -1;
/* both retransmits and lifetime cannot be set */
g_return_val_if_fail ((max_packet_lifetime == -1)
|| (max_retransmits == -1), NULL);
if (!init_params
|| !(protocol = gst_structure_get_string (init_params, "protocol")))
protocol = "";
g_return_val_if_fail (strlen (protocol) <= 65535, NULL);
if (!init_params
|| !gst_structure_get_boolean (init_params, "negotiated", &negotiated))
negotiated = FALSE;
if (!negotiated || !init_params
|| !gst_structure_get_int (init_params, "id", &id))
id = -1;
if (negotiated)
g_return_val_if_fail (id != -1, NULL);
g_return_val_if_fail (id < 65535, NULL);
if (!init_params
|| !gst_structure_get_enum (init_params, "priority",
GST_TYPE_WEBRTC_PRIORITY_TYPE, (gint *) & priority))
priority = GST_WEBRTC_PRIORITY_TYPE_LOW;
/* FIXME: clamp max-retransmits and max-packet-lifetime */
if (webrtc->priv->sctp_transport) {
/* Let transport be the connection's [[SctpTransport]] slot.
*
* If the [[DataChannelId]] slot is not null, transport is in
* connected state and [[DataChannelId]] is greater or equal to the
* transport's [[MaxChannels]] slot, throw an OperationError.
*/
g_object_get (webrtc->priv->sctp_transport, "max-channels", &max_channels,
NULL);
if (max_channels <= 0) {
max_channels = 65534;
}
g_return_val_if_fail (id <= max_channels, NULL);
}
if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc) ||
!_have_sctp_elements (webrtc))
return NULL;
PC_LOCK (webrtc);
DC_LOCK (webrtc);
/* check if the id has been used already */
if (id != -1) {
WebRTCDataChannel *channel = _find_data_channel_for_id (webrtc, id);
if (channel) {
GST_ELEMENT_WARNING (webrtc, LIBRARY, SETTINGS,
("Attempting to add a data channel with a duplicate ID: %i", id),
NULL);
DC_UNLOCK (webrtc);
PC_UNLOCK (webrtc);
return NULL;
}
} else if (webrtc->current_local_description
&& webrtc->current_remote_description && webrtc->priv->sctp_transport
&& webrtc->priv->sctp_transport->transport) {
/* else we can only generate an id if we're configured already. The other
* case for generating an id is on sdp setting */
id = _generate_data_channel_id (webrtc);
if (id == -1) {
GST_ELEMENT_WARNING (webrtc, RESOURCE, NOT_FOUND,
("%s", "Failed to generate an identifier for a data channel"), NULL);
DC_UNLOCK (webrtc);
PC_UNLOCK (webrtc);
return NULL;
}
}
ret = g_object_new (WEBRTC_TYPE_DATA_CHANNEL, "label", label,
"ordered", ordered, "max-packet-lifetime", max_packet_lifetime,
"max-retransmits", max_retransmits, "protocol", protocol,
"negotiated", negotiated, "id", id, "priority", priority, NULL);
if (!ret) {
DC_UNLOCK (webrtc);
PC_UNLOCK (webrtc);
return ret;
}
g_signal_emit (webrtc, gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL], 0,
ret, TRUE);
gst_bin_add (GST_BIN (webrtc), ret->src_bin);
gst_bin_add (GST_BIN (webrtc), ret->sink_bin);
gst_element_sync_state_with_parent (ret->src_bin);
gst_element_sync_state_with_parent (ret->sink_bin);
ret = gst_object_ref (ret);
webrtc_data_channel_set_webrtcbin (ret, webrtc);
g_ptr_array_add (webrtc->priv->data_channels, ret);
webrtc->priv->data_channels_opened++;
DC_UNLOCK (webrtc);
gst_webrtc_bin_update_sctp_priority (webrtc);
webrtc_data_channel_link_to_sctp (ret, webrtc->priv->sctp_transport);
if (webrtc->priv->sctp_transport &&
webrtc->priv->sctp_transport->association_established
&& !ret->parent.negotiated) {
webrtc_data_channel_start_negotiation (ret);
} else {
_update_need_negotiation (webrtc);
}
PC_UNLOCK (webrtc);
return ret;
}
/* === rtpbin signal implementations === */
static void
on_rtpbin_pad_added (GstElement * rtpbin, GstPad * new_pad,
GstWebRTCBin * webrtc)
{
gchar *new_pad_name = NULL;
new_pad_name = gst_pad_get_name (new_pad);
GST_TRACE_OBJECT (webrtc, "new rtpbin pad %s", new_pad_name);
if (g_str_has_prefix (new_pad_name, "recv_rtp_src_")) {
guint32 session_id = 0, ssrc = 0, pt = 0;
SsrcMapItem *mid_entry;
GstWebRTCRTPTransceiver *rtp_trans = NULL;
WebRTCTransceiver *trans;
TransportStream *stream;
GstWebRTCBinPad *pad;
guint media_idx;
if (sscanf (new_pad_name, "recv_rtp_src_%u_%u_%u", &session_id, &ssrc,
&pt) != 3) {
g_critical ("Invalid rtpbin pad name \'%s\'", new_pad_name);
return;
}
media_idx = session_id;
PC_LOCK (webrtc);
stream = _find_transport_for_session (webrtc, session_id);
if (!stream)
g_warn_if_reached ();
mid_entry =
find_mid_ssrc_for_ssrc (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, session_id, ssrc);
if (mid_entry) {
if (mid_entry->mid) {
/* Can't use the mid_entry if the mid doesn't exist */
rtp_trans = _find_transceiver_for_mid (webrtc, mid_entry->mid);
if (rtp_trans) {
g_assert_cmpint (rtp_trans->mline, ==, mid_entry->media_idx);
}
}
if (mid_entry->media_idx != -1)
media_idx = mid_entry->media_idx;
} else {
GST_WARNING_OBJECT (webrtc, "Could not find ssrc %u", ssrc);
/* TODO: connect up to fakesink and reconnect later when this information
* is known from RTCP SDES or RTP Header extension
*/
}
if (!rtp_trans)
rtp_trans = _find_transceiver_for_mline (webrtc, media_idx);
if (!rtp_trans)
g_warn_if_reached ();
trans = WEBRTC_TRANSCEIVER (rtp_trans);
g_assert (trans->stream == stream);
pad = _find_pad_for_transceiver (webrtc, GST_PAD_SRC, rtp_trans);
GST_TRACE_OBJECT (webrtc, "found pad %" GST_PTR_FORMAT
" for rtpbin pad name %s", pad, new_pad_name);
if (!_remove_pending_pad (webrtc, pad)) {
/* assumption here is that rtpbin doesn't duplicate pads and that if
* there is no pending pad, this is a duplicate stream for e.g. simulcast
* or somesuch */
gst_clear_object (&pad);
pad =
_create_pad_for_sdp_media (webrtc, GST_PAD_SRC, rtp_trans, G_MAXUINT,
NULL);
GST_TRACE_OBJECT (webrtc,
"duplicate output ssrc? created new pad %" GST_PTR_FORMAT " for %"
GST_PTR_FORMAT " for rtp pad %s", pad, rtp_trans, new_pad_name);
gst_object_ref_sink (pad);
}
if (!pad)
g_warn_if_reached ();
gst_ghost_pad_set_target (GST_GHOST_PAD (pad), GST_PAD (new_pad));
if (webrtc->priv->running)
gst_pad_set_active (GST_PAD (pad), TRUE);
PC_UNLOCK (webrtc);
gst_pad_sticky_events_foreach (new_pad, copy_sticky_events, pad);
gst_element_add_pad (GST_ELEMENT (webrtc), GST_PAD (pad));
gst_object_unref (pad);
}
g_free (new_pad_name);
}
/* only used for the receiving streams */
static GstCaps *
on_rtpbin_request_pt_map (GstElement * rtpbin, guint session_id, guint pt,
GstWebRTCBin * webrtc)
{
TransportStream *stream;
GstCaps *ret;
GST_DEBUG_OBJECT (webrtc, "getting pt map for pt %d in session %d", pt,
session_id);
PC_LOCK (webrtc);
stream = _find_transport_for_session (webrtc, session_id);
if (!stream)
goto unknown_session;
if ((ret = transport_stream_get_caps_for_pt (stream, pt)))
gst_caps_ref (ret);
GST_DEBUG_OBJECT (webrtc, "Found caps %" GST_PTR_FORMAT " for pt %d in "
"session %d", ret, pt, session_id);
PC_UNLOCK (webrtc);
return ret;
unknown_session:
{
PC_UNLOCK (webrtc);
GST_DEBUG_OBJECT (webrtc, "unknown session %d", session_id);
return NULL;
}
}
static GstElement *
on_rtpbin_request_aux_sender (GstElement * rtpbin, guint session_id,
GstWebRTCBin * webrtc)
{
TransportStream *stream;
GstElement *ret, *rtx;
GstPad *pad;
char *name;
GstElement *aux_sender = NULL;
stream = _find_transport_for_session (webrtc, session_id);
if (!stream) {
/* a rtp session without a stream is a webrtcbin bug */
g_warn_if_reached ();
return NULL;
}
if (stream->rtxsend) {
GST_WARNING_OBJECT (webrtc, "rtprtxsend already created! rtpbin bug?!");
g_warn_if_reached ();
return NULL;
}
GST_DEBUG_OBJECT (webrtc, "requesting aux sender for session %u "
"stream %" GST_PTR_FORMAT, session_id, stream);
ret = gst_bin_new (NULL);
rtx = gst_element_factory_make ("rtprtxsend", NULL);
/* XXX: allow control from outside? */
g_object_set (rtx, "max-size-packets", 500, NULL);
if (!gst_bin_add (GST_BIN (ret), rtx))
g_warn_if_reached ();
ensure_rtx_hdr_ext (stream);
stream->rtxsend = gst_object_ref (rtx);
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
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);
g_clear_pointer (&name, g_free);
name = g_strdup_printf ("sink_%u", session_id);
pad = gst_element_get_static_pad (rtx, "sink");
if (!gst_element_add_pad (ret, gst_ghost_pad_new (name, pad)))
g_warn_if_reached ();
gst_clear_object (&pad);
g_clear_pointer (&name, g_free);
return ret;
}
static GstElement *
on_rtpbin_request_aux_receiver (GstElement * rtpbin, guint session_id,
GstWebRTCBin * webrtc)
{
TransportStream *stream;
GstPad *pad, *ghost;
GstElement *ret;
char *name;
stream = _find_transport_for_session (webrtc, session_id);
if (!stream) {
/* no transport stream before the session has been created is a webrtcbin
* programming error! */
g_warn_if_reached ();
return NULL;
}
if (stream->rtxreceive) {
GST_WARNING_OBJECT (webrtc, "rtprtxreceive already created! rtpbin bug?!");
g_warn_if_reached ();
return NULL;
}
if (stream->reddec) {
GST_WARNING_OBJECT (webrtc, "rtpreddec already created! rtpbin bug?!");
g_warn_if_reached ();
return NULL;
}
GST_DEBUG_OBJECT (webrtc, "requesting aux receiver for session %u "
"stream %" GST_PTR_FORMAT, session_id, stream);
ret = gst_bin_new (NULL);
stream->rtxreceive = gst_element_factory_make ("rtprtxreceive", NULL);
gst_object_ref (stream->rtxreceive);
if (!gst_bin_add (GST_BIN (ret), stream->rtxreceive))
g_warn_if_reached ();
ensure_rtx_hdr_ext (stream);
stream->reddec = gst_element_factory_make ("rtpreddec", NULL);
gst_object_ref (stream->reddec);
if (!gst_bin_add (GST_BIN (ret), stream->reddec))
g_warn_if_reached ();
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
if (!gst_element_link (stream->rtxreceive, stream->reddec))
g_warn_if_reached ();
name = g_strdup_printf ("sink_%u", session_id);
pad = gst_element_get_static_pad (stream->rtxreceive, "sink");
ghost = gst_ghost_pad_new (name, pad);
g_clear_pointer (&name, g_free);
gst_clear_object (&pad);
if (!gst_element_add_pad (ret, ghost))
g_warn_if_reached ();
name = g_strdup_printf ("src_%u", session_id);
pad = gst_element_get_static_pad (stream->reddec, "src");
ghost = gst_ghost_pad_new (name, pad);
g_clear_pointer (&name, g_free);
gst_clear_object (&pad);
if (!gst_element_add_pad (ret, ghost))
g_warn_if_reached ();
return ret;
}
static GstElement *
on_rtpbin_request_fec_decoder_full (GstElement * rtpbin, guint session_id,
guint ssrc, guint pt, GstWebRTCBin * webrtc)
{
TransportStream *stream;
GstElement *ret = NULL;
GObject *internal_storage;
stream = _find_transport_for_session (webrtc, session_id);
if (!stream) {
/* a rtp session without a stream is a webrtcbin bug */
g_warn_if_reached ();
return NULL;
}
/* TODO: for now, we only support ulpfec, but once we support
* more algorithms, if the remote may use more than one algorithm,
* we will want to do the following:
*
* + Return a bin here, with the relevant FEC decoders plugged in
* and their payload type set to 0
*/
GST_DEBUG_OBJECT (webrtc, "Creating ULPFEC decoder for pt %d in session %u "
"stream %" GST_PTR_FORMAT, pt, session_id, stream);
ret = gst_element_factory_make ("rtpulpfecdec", NULL);
g_signal_emit_by_name (webrtc->rtpbin, "get-internal-storage", session_id,
&internal_storage);
g_object_set (ret, "storage", internal_storage, NULL);
g_clear_object (&internal_storage);
g_object_set_data (G_OBJECT (ret), GST_WEBRTC_PAYLOAD_TYPE,
GINT_TO_POINTER (pt));
PC_LOCK (webrtc);
stream->fecdecs = g_list_prepend (stream->fecdecs, gst_object_ref (ret));
_set_internal_rtpbin_element_props_from_stream (webrtc, stream);
PC_UNLOCK (webrtc);
return ret;
}
static void
on_rtpbin_bye_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_INFO_OBJECT (webrtc, "session %u ssrc %u received bye", session_id, ssrc);
PC_LOCK (webrtc);
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
PC_UNLOCK (webrtc);
}
static void
on_rtpbin_bye_timeout (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_INFO_OBJECT (webrtc, "session %u ssrc %u bye timeout", session_id, ssrc);
PC_LOCK (webrtc);
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
PC_UNLOCK (webrtc);
}
static void
on_rtpbin_sender_timeout (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_INFO_OBJECT (webrtc, "session %u ssrc %u sender timeout", session_id,
ssrc);
PC_LOCK (webrtc);
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
PC_UNLOCK (webrtc);
}
static void
on_rtpbin_new_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_INFO_OBJECT (webrtc, "session %u ssrc %u new ssrc", session_id, ssrc);
if (ssrc == 0)
return;
PC_LOCK (webrtc);
find_or_add_ssrc_map_item (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY, session_id, ssrc, -1);
PC_UNLOCK (webrtc);
}
static void
on_rtpbin_ssrc_active (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_TRACE_OBJECT (webrtc, "session %u ssrc %u active", session_id, ssrc);
}
static void
on_rtpbin_ssrc_collision (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_INFO_OBJECT (webrtc, "session %u ssrc %u collision", session_id, ssrc);
}
static void
on_rtpbin_ssrc_sdes (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GObject *session;
GST_INFO_OBJECT (webrtc, "session %u ssrc %u sdes", session_id, ssrc);
g_signal_emit_by_name (rtpbin, "get-internal-session", session_id, &session);
if (session) {
GObject *source;
g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &source);
if (source) {
GstStructure *sdes;
g_object_get (source, "sdes", &sdes, NULL);
/* TODO: when the sdes contains the mid, use that to correlate streams
* as necessary */
GST_DEBUG_OBJECT (webrtc, "session %u ssrc %u sdes %" GST_PTR_FORMAT,
session_id, ssrc, sdes);
gst_clear_structure (&sdes);
gst_clear_object (&source);
}
g_clear_object (&session);
}
}
static void
on_rtpbin_ssrc_validated (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_INFO_OBJECT (webrtc, "session %u ssrc %u validated", session_id, ssrc);
}
static void
on_rtpbin_timeout (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_INFO_OBJECT (webrtc, "session %u ssrc %u timeout", session_id, ssrc);
PC_LOCK (webrtc);
remove_ssrc_entry_by_ssrc (webrtc, session_id, ssrc);
PC_UNLOCK (webrtc);
}
static void
on_rtpbin_new_sender_ssrc (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
SsrcMapItem *mid;
GST_INFO_OBJECT (webrtc, "session %u ssrc %u new sender ssrc", session_id,
ssrc);
PC_LOCK (webrtc);
mid = find_mid_ssrc_for_ssrc (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, session_id, ssrc);
if (!mid) {
TransportStream *stream = _find_transport_for_session (webrtc, session_id);
transport_stream_add_ssrc_map_item (stream,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, ssrc, -1);
} else if (mid->mid) {
/* XXX: when peers support the sdes rtcp item, use this to send the mid rtcp
* sdes item. Requires being able to set the sdes on the rtpsource. */
#if 0
GObject *session;
g_signal_emit_by_name (rtpbin, "get-internal-session", session_id,
&session, NULL);
if (session) {
GObject *source;
g_signal_emit_by_name (session, "get-source-by-ssrc", ssrc, &source);
if (source) {
GstStructure *sdes;
const char *sdes_field_name;
g_object_get (source, "sdes", &sdes, NULL);
GST_WARNING_OBJECT (webrtc, "session %u ssrc %u retrieve sdes %"
GST_PTR_FORMAT, session_id, ssrc, sdes);
sdes_field_name = gst_rtcp_sdes_type_to_name (GST_RTCP_SDES_MID);
g_assert (sdes_field_name);
gst_structure_set (sdes, sdes_field_name, G_TYPE_STRING, mid->mid,
NULL);
if (mid->rid) {
sdes_field_name =
gst_rtcp_sdes_type_to_name (GST_RTCP_SDES_RTP_STREAM_ID);
g_assert (sdes_field_name);
gst_structure_set (sdes, sdes_field_name, mid->rid, NULL);
// TODO: repaired-rtp-stream-id
}
// TODO: writable sdes?
g_object_set (source, "sdes", sdes, NULL);
GST_INFO_OBJECT (webrtc,
"session %u ssrc %u set sdes %" GST_PTR_FORMAT, session_id, ssrc,
sdes);
gst_clear_structure (&sdes);
gst_clear_object (&source);
}
g_clear_object (&session);
}
#endif
}
PC_UNLOCK (webrtc);
}
static void
on_rtpbin_sender_ssrc_active (GstElement * rtpbin, guint session_id, guint ssrc,
GstWebRTCBin * webrtc)
{
GST_TRACE_OBJECT (webrtc, "session %u ssrc %u sender ssrc active", session_id,
ssrc);
}
struct new_jb_args
{
GstWebRTCBin *webrtc;
GstElement *jitterbuffer;
TransportStream *stream;
guint ssrc;
};
static gboolean
jitter_buffer_set_retransmission (SsrcMapItem * item,
const struct new_jb_args *data)
{
GstWebRTCRTPTransceiver *trans;
gboolean do_nack;
if (item->media_idx == -1)
return TRUE;
trans = _find_transceiver_for_mline (data->webrtc, item->media_idx);
if (!trans) {
g_warn_if_reached ();
return TRUE;
}
do_nack = WEBRTC_TRANSCEIVER (trans)->do_nack;
/* We don't set do-retransmission on rtpbin as we want per-session control */
GST_LOG_OBJECT (data->webrtc, "setting do-nack=%s for transceiver %"
GST_PTR_FORMAT " with transport %" GST_PTR_FORMAT
" rtp session %u ssrc %u", do_nack ? "true" : "false", trans,
data->stream, data->stream->session_id, data->ssrc);
g_object_set (data->jitterbuffer, "do-retransmission", do_nack, NULL);
g_weak_ref_set (&item->rtpjitterbuffer, data->jitterbuffer);
return TRUE;
}
static void
on_rtpbin_new_jitterbuffer (GstElement * rtpbin, GstElement * jitterbuffer,
guint session_id, guint ssrc, GstWebRTCBin * webrtc)
{
TransportStream *stream;
struct new_jb_args d = { 0, };
PC_LOCK (webrtc);
GST_INFO_OBJECT (webrtc, "new jitterbuffer %" GST_PTR_FORMAT " for "
"session %u ssrc %u", jitterbuffer, session_id, ssrc);
if (!(stream = _find_transport_for_session (webrtc, session_id))) {
g_warn_if_reached ();
goto out;
}
d.webrtc = webrtc;
d.jitterbuffer = jitterbuffer;
d.stream = stream;
d.ssrc = ssrc;
transport_stream_filter_ssrc_map_item (stream, &d,
(FindSsrcMapFunc) jitter_buffer_set_retransmission);
out:
PC_UNLOCK (webrtc);
}
static void
on_rtpbin_new_storage (GstElement * rtpbin, GstElement * storage,
guint session_id, GstWebRTCBin * webrtc)
{
guint64 latency = webrtc->priv->jb_latency;
/* Add an extra 50 ms for safey */
latency += RTPSTORAGE_EXTRA_TIME;
latency *= GST_MSECOND;
g_object_set (storage, "size-time", latency, NULL);
}
static GstElement *
_create_rtpbin (GstWebRTCBin * webrtc)
{
GstElement *rtpbin;
if (!(rtpbin = gst_element_factory_make ("rtpbin", "rtpbin")))
return NULL;
/* mandated by WebRTC */
gst_util_set_object_arg (G_OBJECT (rtpbin), "rtp-profile", "savpf");
g_object_set (rtpbin, "do-lost", TRUE, NULL);
g_signal_connect (rtpbin, "pad-added", G_CALLBACK (on_rtpbin_pad_added),
webrtc);
g_signal_connect (rtpbin, "request-pt-map",
G_CALLBACK (on_rtpbin_request_pt_map), webrtc);
g_signal_connect (rtpbin, "request-aux-sender",
G_CALLBACK (on_rtpbin_request_aux_sender), webrtc);
g_signal_connect (rtpbin, "request-aux-receiver",
G_CALLBACK (on_rtpbin_request_aux_receiver), webrtc);
g_signal_connect (rtpbin, "new-storage",
G_CALLBACK (on_rtpbin_new_storage), webrtc);
g_signal_connect (rtpbin, "request-fec-decoder-full",
G_CALLBACK (on_rtpbin_request_fec_decoder_full), webrtc);
g_signal_connect (rtpbin, "on-bye-ssrc",
G_CALLBACK (on_rtpbin_bye_ssrc), webrtc);
g_signal_connect (rtpbin, "on-bye-timeout",
G_CALLBACK (on_rtpbin_bye_timeout), webrtc);
g_signal_connect (rtpbin, "on-new-ssrc",
G_CALLBACK (on_rtpbin_new_ssrc), webrtc);
g_signal_connect (rtpbin, "on-new-sender-ssrc",
G_CALLBACK (on_rtpbin_new_sender_ssrc), webrtc);
g_signal_connect (rtpbin, "on-sender-ssrc-active",
G_CALLBACK (on_rtpbin_sender_ssrc_active), webrtc);
g_signal_connect (rtpbin, "on-sender-timeout",
G_CALLBACK (on_rtpbin_sender_timeout), webrtc);
g_signal_connect (rtpbin, "on-ssrc-active",
G_CALLBACK (on_rtpbin_ssrc_active), webrtc);
g_signal_connect (rtpbin, "on-ssrc-collision",
G_CALLBACK (on_rtpbin_ssrc_collision), webrtc);
g_signal_connect (rtpbin, "on-ssrc-sdes",
G_CALLBACK (on_rtpbin_ssrc_sdes), webrtc);
g_signal_connect (rtpbin, "on-ssrc-validated",
G_CALLBACK (on_rtpbin_ssrc_validated), webrtc);
g_signal_connect (rtpbin, "on-timeout",
G_CALLBACK (on_rtpbin_timeout), webrtc);
g_signal_connect (rtpbin, "new-jitterbuffer",
G_CALLBACK (on_rtpbin_new_jitterbuffer), webrtc);
return rtpbin;
}
static GstStateChangeReturn
gst_webrtc_bin_change_state (GstElement * element, GstStateChange transition)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GST_DEBUG ("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:{
if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc))
return GST_STATE_CHANGE_FAILURE;
_start_thread (webrtc);
PC_LOCK (webrtc);
_update_need_negotiation (webrtc);
PC_UNLOCK (webrtc);
break;
}
case GST_STATE_CHANGE_READY_TO_PAUSED:
webrtc->priv->running = TRUE;
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_PAUSED:
/* Mangle the return value to NO_PREROLL as that's what really is
* occurring here however cannot be propagated correctly due to nicesrc
* requiring that it be in PLAYING already in order to send/receive
* correctly :/ */
ret = GST_STATE_CHANGE_NO_PREROLL;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
webrtc->priv->running = FALSE;
break;
case GST_STATE_CHANGE_READY_TO_NULL:
_stop_thread (webrtc);
break;
default:
break;
}
return ret;
}
static GstPadProbeReturn
sink_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 void
peek_sink_buffer (GstWebRTCBin * webrtc, guint rtp_session_id,
guint media_idx, WebRTCTransceiver * trans, GstBuffer * buffer)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
SsrcMapItem *item;
guint ssrc;
if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
return;
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
gst_rtp_buffer_unmap (&rtp);
if (!ssrc) {
GST_WARNING_OBJECT (webrtc,
"incoming buffer does not contain a valid ssrc");
return;
}
PC_LOCK (webrtc);
item =
find_or_add_ssrc_map_item (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY, rtp_session_id, ssrc,
media_idx);
if (item->media_idx == -1) {
char *str;
GST_DEBUG_OBJECT (webrtc, "updating media idx of ssrc item %p to %u", item,
media_idx);
item->media_idx = media_idx;
/* ensure that the rtx mapping contains a valid ssrc to use for rtx when
* used even when there are no ssrc's in the input/codec preferences caps */
str = g_strdup_printf ("%u", ssrc);
if (!gst_structure_has_field_typed (trans->local_rtx_ssrc_map, str,
G_TYPE_UINT)) {
/* TODO: ssrc-collision? */
gst_structure_set (trans->local_rtx_ssrc_map, str, G_TYPE_UINT,
g_random_int (), NULL);
_set_internal_rtpbin_element_props_from_stream (webrtc, trans->stream);
}
g_free (str);
}
PC_UNLOCK (webrtc);
}
static GstPadProbeReturn
sink_pad_buffer_peek (GstPad * pad, GstPadProbeInfo * info,
GstWebRTCBin * webrtc)
{
GstWebRTCBinPad *webrtc_pad = GST_WEBRTC_BIN_PAD (pad);
WebRTCTransceiver *trans;
guint rtp_session_id, media_idx;
if (!webrtc_pad->trans)
return GST_PAD_PROBE_OK;
trans = (WebRTCTransceiver *) webrtc_pad->trans;
if (!trans->stream)
return GST_PAD_PROBE_OK;
rtp_session_id = trans->stream->session_id;
media_idx = webrtc_pad->trans->mline;
if (media_idx != G_MAXUINT)
return GST_PAD_PROBE_OK;
if (info->type & GST_PAD_PROBE_TYPE_BUFFER) {
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
peek_sink_buffer (webrtc, rtp_session_id, media_idx, trans, buffer);
} else if (info->type & GST_PAD_PROBE_TYPE_BUFFER_LIST) {
GstBufferList *list = GST_PAD_PROBE_INFO_BUFFER_LIST (info);
guint i, n;
n = gst_buffer_list_length (list);
for (i = 0; i < n; i++) {
GstBuffer *buffer = gst_buffer_list_get (list, i);
peek_sink_buffer (webrtc, rtp_session_id, media_idx, trans, buffer);
}
} else {
g_assert_not_reached ();
}
return GST_PAD_PROBE_OK;
}
static GstPad *
gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
const gchar * name, const GstCaps * caps)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
GstWebRTCRTPTransceiver *trans = NULL, *created_trans = NULL;
GstWebRTCBinPad *pad = NULL;
guint serial;
gboolean lock_mline = FALSE;
if (!_have_nice_elements (webrtc) || !_have_dtls_elements (webrtc))
return NULL;
if (templ->direction != GST_PAD_SINK ||
g_strcmp0 (templ->name_template, "sink_%u") != 0) {
GST_ERROR_OBJECT (element, "Requested pad that shouldn't be requestable");
return NULL;
}
PC_LOCK (webrtc);
if (name == NULL || strlen (name) < 6 || !g_str_has_prefix (name, "sink_")) {
/* no name given when requesting the pad, use next available int */
serial = webrtc->priv->max_sink_pad_serial++;
} else {
/* parse serial number from requested padname */
serial = g_ascii_strtoull (&name[5], NULL, 10);
lock_mline = TRUE;
}
if (lock_mline) {
GstWebRTCBinPad *pad2;
trans = _find_transceiver_for_mline (webrtc, serial);
if (trans) {
/* Reject transceivers that are only for receiving ... */
if (trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY ||
trans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE) {
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
" existing m-line %d, but the transceiver's direction is %s",
name, serial,
gst_webrtc_rtp_transceiver_direction_to_string (trans->direction));
goto error_out;
}
/* Reject transceivers that already have a pad allocated */
pad2 = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, trans);
if (pad2) {
GST_ERROR_OBJECT (element, "Trying to request pad %s for m-line %d, "
" but the transceiver associated with this m-line already has pad"
" %s", name, serial, GST_PAD_NAME (pad2));
gst_object_unref (pad2);
goto error_out;
}
if (caps) {
GST_OBJECT_LOCK (trans);
if (trans->codec_preferences &&
!gst_caps_can_intersect (caps, trans->codec_preferences)) {
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
" existing m-line %d, but requested caps %" GST_PTR_FORMAT
" don't match existing codec preferences %" GST_PTR_FORMAT,
name, serial, caps, trans->codec_preferences);
GST_OBJECT_UNLOCK (trans);
goto error_out;
}
GST_OBJECT_UNLOCK (trans);
if (trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
GstWebRTCKind kind = webrtc_kind_from_caps (caps);
if (trans->kind != kind) {
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
" existing m-line %d, but requested caps %" GST_PTR_FORMAT
" don't match transceiver kind %d",
name, serial, caps, trans->kind);
goto error_out;
}
}
}
}
}
/* Let's try to find a free transceiver that matches */
if (!trans) {
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
guint i;
kind = webrtc_kind_from_caps (caps);
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
GstWebRTCRTPTransceiver *tmptrans =
g_ptr_array_index (webrtc->priv->transceivers, i);
GstWebRTCBinPad *pad2;
gboolean has_matching_caps;
/* Ignore transceivers with a non-matching kind */
if (tmptrans->kind != GST_WEBRTC_KIND_UNKNOWN &&
kind != GST_WEBRTC_KIND_UNKNOWN && tmptrans->kind != kind)
continue;
/* Ignore stopped transmitters */
if (tmptrans->stopped)
continue;
/* Ignore transceivers that are only for receiving ... */
if (tmptrans->direction == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_RECVONLY
|| tmptrans->direction ==
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_INACTIVE)
continue;
/* Ignore transceivers that already have a pad allocated */
pad2 = _find_pad_for_transceiver (webrtc, GST_PAD_SINK, tmptrans);
if (pad2) {
gst_object_unref (pad2);
continue;
}
GST_OBJECT_LOCK (tmptrans);
has_matching_caps = (caps && tmptrans->codec_preferences &&
!gst_caps_can_intersect (caps, tmptrans->codec_preferences));
GST_OBJECT_UNLOCK (tmptrans);
/* Ignore transceivers with non-matching caps */
if (!has_matching_caps)
continue;
trans = tmptrans;
break;
}
}
if (!trans) {
trans = created_trans =
GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, -1,
webrtc_kind_from_caps (caps), NULL));
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT, trans);
} else {
GST_LOG_OBJECT (webrtc, "Using existing transceiver %" GST_PTR_FORMAT
" for mline %u", trans, serial);
if (caps) {
if (!_update_transceiver_kind_from_caps (trans, caps)) {
GstWebRTCKind caps_kind = webrtc_kind_from_caps (caps);
GST_WARNING_OBJECT (webrtc,
"Trying to change kind of transceiver %" GST_PTR_FORMAT
" at m-line %d from %s (%d) to %s (%d)", trans, serial,
gst_webrtc_kind_to_string (trans->kind), trans->kind,
gst_webrtc_kind_to_string (caps_kind), caps_kind);
}
}
}
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, trans, serial, NULL);
pad->block_id = gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BLOCK |
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
(GstPadProbeCallback) sink_pad_block, NULL, NULL);
webrtc->priv->pending_sink_transceivers =
g_list_append (webrtc->priv->pending_sink_transceivers,
gst_object_ref (pad));
gst_pad_add_probe (GST_PAD (pad),
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
(GstPadProbeCallback) sink_pad_buffer_peek, webrtc, NULL);
if (lock_mline) {
WebRTCTransceiver *wtrans = WEBRTC_TRANSCEIVER (trans);
wtrans->mline_locked = TRUE;
trans->mline = serial;
}
PC_UNLOCK (webrtc);
if (created_trans)
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL],
0, created_trans);
_add_pad (webrtc, pad);
return GST_PAD (pad);
error_out:
PC_UNLOCK (webrtc);
return NULL;
}
static void
gst_webrtc_bin_release_pad (GstElement * element, GstPad * pad)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (element);
GstWebRTCBinPad *webrtc_pad = GST_WEBRTC_BIN_PAD (pad);
GST_DEBUG_OBJECT (webrtc, "Releasing %" GST_PTR_FORMAT, webrtc_pad);
/* remove the transceiver from the pad so that subsequent code doesn't use
* a possibly dead transceiver */
PC_LOCK (webrtc);
if (webrtc_pad->trans)
gst_object_unref (webrtc_pad->trans);
webrtc_pad->trans = NULL;
gst_caps_replace (&webrtc_pad->received_caps, NULL);
PC_UNLOCK (webrtc);
if (webrtc_pad->block_id) {
gst_pad_remove_probe (GST_PAD (pad), webrtc_pad->block_id);
webrtc_pad->block_id = 0;
}
_remove_pad (webrtc, webrtc_pad);
PC_LOCK (webrtc);
_update_need_negotiation (webrtc);
PC_UNLOCK (webrtc);
}
static void
_update_rtpstorage_latency (GstWebRTCBin * webrtc)
{
guint i;
guint64 latency_ns;
/* Add an extra 50 ms for safety */
latency_ns = webrtc->priv->jb_latency + RTPSTORAGE_EXTRA_TIME;
latency_ns *= GST_MSECOND;
for (i = 0; i < webrtc->priv->transports->len; i++) {
TransportStream *stream = g_ptr_array_index (webrtc->priv->transports, i);
GObject *storage = NULL;
g_signal_emit_by_name (webrtc->rtpbin, "get-storage", stream->session_id,
&storage);
g_object_set (storage, "size-time", latency_ns, NULL);
g_object_unref (storage);
}
}
static void
gst_webrtc_bin_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
switch (prop_id) {
case PROP_STUN_SERVER:
gst_webrtc_ice_set_stun_server (webrtc->priv->ice,
g_value_get_string (value));
break;
case PROP_TURN_SERVER:
gst_webrtc_ice_set_turn_server (webrtc->priv->ice,
g_value_get_string (value));
break;
case PROP_BUNDLE_POLICY:
if (g_value_get_enum (value) == GST_WEBRTC_BUNDLE_POLICY_BALANCED) {
GST_ERROR_OBJECT (object, "Balanced bundle policy not implemented yet");
} else {
webrtc->bundle_policy = g_value_get_enum (value);
}
break;
case PROP_ICE_TRANSPORT_POLICY:
webrtc->ice_transport_policy = g_value_get_enum (value);
gst_webrtc_ice_set_force_relay (webrtc->priv->ice,
webrtc->ice_transport_policy ==
GST_WEBRTC_ICE_TRANSPORT_POLICY_RELAY ? TRUE : FALSE);
break;
case PROP_LATENCY:
g_object_set_property (G_OBJECT (webrtc->rtpbin), "latency", value);
webrtc->priv->jb_latency = g_value_get_uint (value);
_update_rtpstorage_latency (webrtc);
break;
case PROP_ICE_AGENT:
webrtc->priv->ice = g_value_get_object (value);
break;
case PROP_HTTP_PROXY:
gst_webrtc_ice_set_http_proxy (webrtc->priv->ice,
g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_webrtc_bin_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
PC_LOCK (webrtc);
switch (prop_id) {
case PROP_CONNECTION_STATE:
g_value_set_enum (value, webrtc->peer_connection_state);
break;
case PROP_SIGNALING_STATE:
g_value_set_enum (value, webrtc->signaling_state);
break;
case PROP_ICE_GATHERING_STATE:
g_value_set_enum (value, webrtc->ice_gathering_state);
break;
case PROP_ICE_CONNECTION_STATE:
g_value_set_enum (value, webrtc->ice_connection_state);
break;
case PROP_LOCAL_DESCRIPTION:
if (webrtc->pending_local_description)
g_value_set_boxed (value, webrtc->pending_local_description);
else if (webrtc->current_local_description)
g_value_set_boxed (value, webrtc->current_local_description);
else
g_value_set_boxed (value, NULL);
break;
case PROP_CURRENT_LOCAL_DESCRIPTION:
g_value_set_boxed (value, webrtc->current_local_description);
break;
case PROP_PENDING_LOCAL_DESCRIPTION:
g_value_set_boxed (value, webrtc->pending_local_description);
break;
case PROP_REMOTE_DESCRIPTION:
if (webrtc->pending_remote_description)
g_value_set_boxed (value, webrtc->pending_remote_description);
else if (webrtc->current_remote_description)
g_value_set_boxed (value, webrtc->current_remote_description);
else
g_value_set_boxed (value, NULL);
break;
case PROP_CURRENT_REMOTE_DESCRIPTION:
g_value_set_boxed (value, webrtc->current_remote_description);
break;
case PROP_PENDING_REMOTE_DESCRIPTION:
g_value_set_boxed (value, webrtc->pending_remote_description);
break;
case PROP_STUN_SERVER:
g_value_take_string (value,
gst_webrtc_ice_get_stun_server (webrtc->priv->ice));
break;
case PROP_TURN_SERVER:
g_value_take_string (value,
gst_webrtc_ice_get_turn_server (webrtc->priv->ice));
break;
case PROP_BUNDLE_POLICY:
g_value_set_enum (value, webrtc->bundle_policy);
break;
case PROP_ICE_TRANSPORT_POLICY:
g_value_set_enum (value, webrtc->ice_transport_policy);
break;
case PROP_ICE_AGENT:
g_value_set_object (value, webrtc->priv->ice);
break;
case PROP_LATENCY:
g_value_set_uint (value, webrtc->priv->jb_latency);
break;
case PROP_SCTP_TRANSPORT:
g_value_set_object (value, webrtc->priv->sctp_transport);
break;
case PROP_HTTP_PROXY:
g_value_take_string (value,
gst_webrtc_ice_get_http_proxy (webrtc->priv->ice));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
PC_UNLOCK (webrtc);
}
static void
gst_webrtc_bin_constructed (GObject * object)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
gchar *name;
if (!webrtc->priv->ice) {
name = g_strdup_printf ("%s:ice", GST_OBJECT_NAME (webrtc));
webrtc->priv->ice = GST_WEBRTC_ICE (gst_webrtc_nice_new (name));
g_free (name);
}
gst_webrtc_ice_set_on_ice_candidate (webrtc->priv->ice,
(GstWebRTCICEOnCandidateFunc) _on_local_ice_candidate_cb, webrtc, NULL);
G_OBJECT_CLASS (parent_class)->constructed (object);
}
static void
_free_pending_pad (GstPad * pad)
{
gst_object_unref (pad);
}
static void
gst_webrtc_bin_dispose (GObject * object)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
if (webrtc->priv->ice)
gst_object_unref (webrtc->priv->ice);
webrtc->priv->ice = NULL;
if (webrtc->priv->ice_stream_map)
g_array_free (webrtc->priv->ice_stream_map, TRUE);
webrtc->priv->ice_stream_map = NULL;
g_clear_object (&webrtc->priv->sctp_transport);
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_webrtc_bin_finalize (GObject * object)
{
GstWebRTCBin *webrtc = GST_WEBRTC_BIN (object);
if (webrtc->priv->transports)
g_ptr_array_free (webrtc->priv->transports, TRUE);
webrtc->priv->transports = NULL;
if (webrtc->priv->transceivers)
g_ptr_array_free (webrtc->priv->transceivers, TRUE);
webrtc->priv->transceivers = NULL;
if (webrtc->priv->data_channels)
g_ptr_array_free (webrtc->priv->data_channels, TRUE);
webrtc->priv->data_channels = NULL;
if (webrtc->priv->pending_data_channels)
g_ptr_array_free (webrtc->priv->pending_data_channels, TRUE);
webrtc->priv->pending_data_channels = NULL;
if (webrtc->priv->pending_remote_ice_candidates)
g_array_free (webrtc->priv->pending_remote_ice_candidates, TRUE);
webrtc->priv->pending_remote_ice_candidates = NULL;
if (webrtc->priv->pending_local_ice_candidates)
g_array_free (webrtc->priv->pending_local_ice_candidates, TRUE);
webrtc->priv->pending_local_ice_candidates = NULL;
if (webrtc->priv->pending_pads)
g_list_free_full (webrtc->priv->pending_pads,
(GDestroyNotify) _free_pending_pad);
webrtc->priv->pending_pads = NULL;
if (webrtc->priv->pending_sink_transceivers)
g_list_free_full (webrtc->priv->pending_sink_transceivers,
(GDestroyNotify) gst_object_unref);
webrtc->priv->pending_sink_transceivers = NULL;
if (webrtc->current_local_description)
gst_webrtc_session_description_free (webrtc->current_local_description);
webrtc->current_local_description = NULL;
if (webrtc->pending_local_description)
gst_webrtc_session_description_free (webrtc->pending_local_description);
webrtc->pending_local_description = NULL;
if (webrtc->current_remote_description)
gst_webrtc_session_description_free (webrtc->current_remote_description);
webrtc->current_remote_description = NULL;
if (webrtc->pending_remote_description)
gst_webrtc_session_description_free (webrtc->pending_remote_description);
webrtc->pending_remote_description = NULL;
if (webrtc->priv->last_generated_answer)
gst_webrtc_session_description_free (webrtc->priv->last_generated_answer);
webrtc->priv->last_generated_answer = NULL;
if (webrtc->priv->last_generated_offer)
gst_webrtc_session_description_free (webrtc->priv->last_generated_offer);
webrtc->priv->last_generated_offer = NULL;
g_mutex_clear (DC_GET_LOCK (webrtc));
g_mutex_clear (ICE_GET_LOCK (webrtc));
g_mutex_clear (PC_GET_LOCK (webrtc));
g_cond_clear (PC_GET_COND (webrtc));
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *element_class = (GstElementClass *) klass;
element_class->request_new_pad = gst_webrtc_bin_request_new_pad;
element_class->release_pad = gst_webrtc_bin_release_pad;
element_class->change_state = gst_webrtc_bin_change_state;
gst_element_class_add_static_pad_template_with_gtype (element_class,
&sink_template, GST_TYPE_WEBRTC_BIN_SINK_PAD);
gst_element_class_add_static_pad_template_with_gtype (element_class,
&src_template, GST_TYPE_WEBRTC_BIN_SRC_PAD);
gst_element_class_set_metadata (element_class, "WebRTC Bin",
"Filter/Network/WebRTC", "A bin for webrtc connections",
"Matthew Waters <matthew@centricular.com>");
gobject_class->constructed = gst_webrtc_bin_constructed;
gobject_class->get_property = gst_webrtc_bin_get_property;
gobject_class->set_property = gst_webrtc_bin_set_property;
gobject_class->dispose = gst_webrtc_bin_dispose;
gobject_class->finalize = gst_webrtc_bin_finalize;
g_object_class_install_property (gobject_class,
PROP_LOCAL_DESCRIPTION,
g_param_spec_boxed ("local-description", "Local Description",
"The local SDP description in use for this connection. "
"Favours a pending description over the current description",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_CURRENT_LOCAL_DESCRIPTION,
g_param_spec_boxed ("current-local-description",
"Current Local Description",
"The local description that was successfully negotiated the last time "
"the connection transitioned into the stable state",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_PENDING_LOCAL_DESCRIPTION,
g_param_spec_boxed ("pending-local-description",
"Pending Local Description",
"The local description that is in the process of being negotiated plus "
"any local candidates that have been generated by the ICE Agent since the "
"offer or answer was created",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_REMOTE_DESCRIPTION,
g_param_spec_boxed ("remote-description", "Remote Description",
"The remote SDP description to use for this connection. "
"Favours a pending description over the current description",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_CURRENT_REMOTE_DESCRIPTION,
g_param_spec_boxed ("current-remote-description",
"Current Remote Description",
"The last remote description that was successfully negotiated the last "
"time the connection transitioned into the stable state plus any remote "
"candidates that have been supplied via addIceCandidate() since the offer "
"or answer was created",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_PENDING_REMOTE_DESCRIPTION,
g_param_spec_boxed ("pending-remote-description",
"Pending Remote Description",
"The remote description that is in the process of being negotiated, "
"complete with any remote candidates that have been supplied via "
"addIceCandidate() since the offer or answer was created",
GST_TYPE_WEBRTC_SESSION_DESCRIPTION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_STUN_SERVER,
g_param_spec_string ("stun-server", "STUN Server",
"The STUN server of the form stun://hostname:port",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_TURN_SERVER,
g_param_spec_string ("turn-server", "TURN Server",
"The TURN server of the form turn(s)://username:password@host:port. "
"To use time-limited credentials, the form must be turn(s)://timestamp:"
"username:password@host:port. Please note that the ':' character of "
"the 'timestamp:username' and the 'password' encoded by base64 should "
"be escaped to be parsed properly. "
"This is a convenience property, use #GstWebRTCBin::add-turn-server "
"if you wish to use multiple TURN servers",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_CONNECTION_STATE,
g_param_spec_enum ("connection-state", "Connection State",
"The overall connection state of this element",
GST_TYPE_WEBRTC_PEER_CONNECTION_STATE,
GST_WEBRTC_PEER_CONNECTION_STATE_NEW,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_SIGNALING_STATE,
g_param_spec_enum ("signaling-state", "Signaling State",
"The signaling state of this element",
GST_TYPE_WEBRTC_SIGNALING_STATE,
GST_WEBRTC_SIGNALING_STATE_STABLE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ICE_CONNECTION_STATE,
g_param_spec_enum ("ice-connection-state", "ICE connection state",
"The collective connection state of all ICETransport's",
GST_TYPE_WEBRTC_ICE_CONNECTION_STATE,
GST_WEBRTC_ICE_CONNECTION_STATE_NEW,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ICE_GATHERING_STATE,
g_param_spec_enum ("ice-gathering-state", "ICE gathering state",
"The collective gathering state of all ICETransport's",
GST_TYPE_WEBRTC_ICE_GATHERING_STATE,
GST_WEBRTC_ICE_GATHERING_STATE_NEW,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_BUNDLE_POLICY,
g_param_spec_enum ("bundle-policy", "Bundle Policy",
"The policy to apply for bundling",
GST_TYPE_WEBRTC_BUNDLE_POLICY,
GST_WEBRTC_BUNDLE_POLICY_NONE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ICE_TRANSPORT_POLICY,
g_param_spec_enum ("ice-transport-policy", "ICE Transport Policy",
"The policy to apply for ICE transport",
GST_TYPE_WEBRTC_ICE_TRANSPORT_POLICY,
GST_WEBRTC_ICE_TRANSPORT_POLICY_ALL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ICE_AGENT,
g_param_spec_object ("ice-agent", "WebRTC ICE agent",
"The WebRTC ICE agent",
GST_TYPE_WEBRTC_ICE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT_ONLY));
/**
* GstWebRTCBin:latency:
*
* Default duration to buffer in the jitterbuffers (in ms)
*
* Since: 1.18
*/
g_object_class_install_property (gobject_class,
PROP_LATENCY,
g_param_spec_uint ("latency", "Latency",
"Default duration to buffer in the jitterbuffers (in ms)",
0, G_MAXUINT, DEFAULT_JB_LATENCY,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTCBin:http-proxy:
*
* A HTTP proxy for use with TURN/TCP of the form
* http://[username:password@]hostname[:port][?alpn=<alpn>]
*
* Since: 1.22
*/
g_object_class_install_property (gobject_class,
PROP_HTTP_PROXY,
g_param_spec_string ("http-proxy", "HTTP Proxy",
"A HTTP proxy for use with TURN/TCP of the form "
"http://[username:password@]hostname[:port][?alpn=<alpn>]",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTCBin:sctp-transport:
*
* The WebRTC SCTP Transport
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class,
PROP_SCTP_TRANSPORT,
g_param_spec_object ("sctp-transport", "WebRTC SCTP Transport",
"The WebRTC SCTP Transport",
GST_TYPE_WEBRTC_SCTP_TRANSPORT,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTCBin::create-offer:
* @object: the #webrtcbin
* @options: (nullable): create-offer options
* @promise: a #GstPromise which will contain the offer
*/
gst_webrtc_bin_signals[CREATE_OFFER_SIGNAL] =
g_signal_new_class_handler ("create-offer", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_create_offer), NULL, NULL, NULL,
G_TYPE_NONE, 2, GST_TYPE_STRUCTURE, GST_TYPE_PROMISE);
/**
* GstWebRTCBin::create-answer:
* @object: the #webrtcbin
* @options: (nullable): create-answer options
* @promise: a #GstPromise which will contain the answer
*/
gst_webrtc_bin_signals[CREATE_ANSWER_SIGNAL] =
g_signal_new_class_handler ("create-answer", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_create_answer), NULL, NULL, NULL,
G_TYPE_NONE, 2, GST_TYPE_STRUCTURE, GST_TYPE_PROMISE);
/**
* GstWebRTCBin::set-local-description:
* @object: the #GstWebRTCBin
* @desc: a #GstWebRTCSessionDescription description
* @promise: (nullable): a #GstPromise to be notified when it's set
*/
gst_webrtc_bin_signals[SET_LOCAL_DESCRIPTION_SIGNAL] =
g_signal_new_class_handler ("set-local-description",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_set_local_description), NULL, NULL, NULL,
G_TYPE_NONE, 2, GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE);
/**
* GstWebRTCBin::set-remote-description:
* @object: the #GstWebRTCBin
* @desc: a #GstWebRTCSessionDescription description
* @promise: (nullable): a #GstPromise to be notified when it's set
*/
gst_webrtc_bin_signals[SET_REMOTE_DESCRIPTION_SIGNAL] =
g_signal_new_class_handler ("set-remote-description",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_set_remote_description), NULL, NULL, NULL,
G_TYPE_NONE, 2, GST_TYPE_WEBRTC_SESSION_DESCRIPTION, GST_TYPE_PROMISE);
/**
* GstWebRTCBin::add-ice-candidate:
* @object: the #webrtcbin
* @mline_index: the index of the media description in the SDP
* @ice-candidate: an ice candidate or NULL/"" to mark that no more candidates
* will arrive
*/
gst_webrtc_bin_signals[ADD_ICE_CANDIDATE_SIGNAL] =
g_signal_new_class_handler ("add-ice-candidate",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_add_ice_candidate), NULL, NULL, NULL,
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING);
/**
* GstWebRTCBin::add-ice-candidate-full:
* @object: the #webrtcbin
* @mline_index: the index of the media description in the SDP
* @ice-candidate: an ice candidate or NULL/"" to mark that no more candidates
* will arrive
* @promise: (nullable): a #GstPromise to be notified when the task is
* complete
*
* Variant of the `add-ice-candidate` signal, allowing the call site to be
* notified using a #GstPromise when the task has completed.
*
* Since: 1.24
*/
gst_webrtc_bin_signals[ADD_ICE_CANDIDATE_FULL_SIGNAL] =
g_signal_new_class_handler ("add-ice-candidate-full",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_add_ice_candidate), NULL, NULL, NULL,
G_TYPE_NONE, 3, G_TYPE_UINT, G_TYPE_STRING, GST_TYPE_PROMISE);
/**
* GstWebRTCBin::get-stats:
* @object: the #webrtcbin
* @pad: (nullable): A #GstPad to get the stats for, or %NULL for all
* @promise: a #GstPromise for the result
*
* The @promise will contain the result of retrieving the session statistics.
* The structure will be named 'application/x-webrtc-stats and contain the
* following based on the webrtc-stats spec available from
* https://www.w3.org/TR/webrtc-stats/. As the webrtc-stats spec is a draft
* and is constantly changing these statistics may be changed to fit with
* the latest spec.
*
* Each field key is a unique identifier for each RTCStats
* (https://www.w3.org/TR/webrtc/#rtcstats-dictionary) value (another
* GstStructure) in the RTCStatsReport
* (https://www.w3.org/TR/webrtc/#rtcstatsreport-object). Each supported
* field in the RTCStats subclass is outlined below.
*
* Each statistics structure contains the following values as defined by
* the RTCStats dictionary (https://www.w3.org/TR/webrtc/#rtcstats-dictionary).
*
* "timestamp" G_TYPE_DOUBLE timestamp the statistics were generated
* "type" GST_TYPE_WEBRTC_STATS_TYPE the type of statistics reported
* "id" G_TYPE_STRING unique identifier
*
* RTCCodecStats supported fields (https://w3c.github.io/webrtc-stats/#codec-dict*)
*
* "payload-type" G_TYPE_UINT the rtp payload number in use
* "clock-rate" G_TYPE_UINT the rtp clock-rate
*
* RTCRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#streamstats-dict*)
*
* "ssrc" G_TYPE_STRING the rtp sequence src in use
* "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream
* "codec-id" G_TYPE_STRING identifier for the associated RTCCodecStats for this stream
* "kind" G_TYPE_STRING either "audio" or "video", depending on the associated transceiver (Since: 1.22)
*
* RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)
*
* "packets-received" G_TYPE_UINT64 number of packets received (only for local inbound)
* "packets-lost" G_TYPE_INT64 number of packets lost
* "packets-discarded" G_TYPE_UINT64 number of packets discarded
* "packets-repaired" G_TYPE_UINT64 number of packets repaired
* "jitter" G_TYPE_DOUBLE packet jitter measured in seconds
*
* RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*)
*
* "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteOutboundRTPStreamStats
* "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound)
* "packets-duplicated" G_TYPE_UINT64 number of packets duplicated
* "fir-count" G_TYPE_UINT FIR packets sent by the receiver
* "pli-count" G_TYPE_UINT PLI packets sent by the receiver
* "nack-count" G_TYPE_UINT NACK packets sent by the receiver
*
* RTCRemoteInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*)
*
* "local-id" G_TYPE_STRING identifier for the associated RTCOutboundRTPSTreamStats
* "round-trip-time" G_TYPE_DOUBLE round trip time of packets measured in seconds
* "fraction-lost" G_TYPE_DOUBLE fraction packet loss
*
* RTCSentRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*)
*
* "packets-sent" G_TYPE_UINT64 number of packets sent (only for local outbound)
* "bytes-sent" G_TYPE_UINT64 number of packets sent (only for local outbound)
*
* RTCOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*)
*
* "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteInboundRTPSTreamStats (optional since 1.22)
* "fir-count" G_TYPE_UINT FIR packets received by the sender
* "pli-count" G_TYPE_UINT PLI packets received by the sender
* "nack-count" G_TYPE_UINT NACK packets received by the sender
*
* RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*)
*
* "local-id" G_TYPE_STRING identifier for the associated RTCInboundRTPSTreamStats
* "remote-timestamp" G_TYPE_DOUBLE remote timestamp the statistics were sent by the remote
*
* RTCPeerConnectionStats supported fields (https://w3c.github.io/webrtc-stats/#pcstats-dict*) (Since: 1.24)
*
* "data-channels-opened" G_TYPE_UINT number of unique data channels that have entered the 'open' state
* "data-channels-closed" G_TYPE_UINT number of unique data channels that have left the 'open' state
*
* RTCIceCandidateStats supported fields (https://www.w3.org/TR/webrtc-stats/#icecandidate-dict*) (Since: 1.22)
*
* "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream
* "address" G_TYPE_STRING address of the candidate, allowing for IPv4, IPv6 and FQDNs
* "port" G_TYPE_UINT port number of the candidate
* "candidate-type" G_TYPE_STRING RTCIceCandidateType
* "priority" G_TYPE_UINT64 calculated as defined in RFC 5245
* "protocol" G_TYPE_STRING Either "udp" or "tcp". Based on the "transport" defined in RFC 5245
* "relay-protocol" G_TYPE_STRING protocol used by the endpoint to communicate with the TURN server. Only present for local candidates. Either "udp", "tcp" or "tls"
* "url" G_TYPE_STRING URL of the ICE server from which the candidate was obtained. Only present for local candidates
*
* RTCIceCandidatePairStats supported fields (https://www.w3.org/TR/webrtc-stats/#candidatepair-dict*) (Since: 1.22)
*
* "local-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the local candidate associated with this candidate pair.
* "remote-candidate-id" G_TYPE_STRING unique identifier that is associated to the object that was inspected to produce the RTCIceCandidateStats for the remote candidate associated with this candidate pair.
*/
gst_webrtc_bin_signals[GET_STATS_SIGNAL] =
g_signal_new_class_handler ("get-stats",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_get_stats), NULL, NULL, NULL,
G_TYPE_NONE, 2, GST_TYPE_PAD, GST_TYPE_PROMISE);
/**
* GstWebRTCBin::on-negotiation-needed:
* @object: the #webrtcbin
*/
gst_webrtc_bin_signals[ON_NEGOTIATION_NEEDED_SIGNAL] =
g_signal_new ("on-negotiation-needed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, G_TYPE_NONE, 0);
/**
* GstWebRTCBin::on-ice-candidate:
* @object: the #webrtcbin
* @mline_index: the index of the media description in the SDP
* @candidate: the ICE candidate
*/
gst_webrtc_bin_signals[ON_ICE_CANDIDATE_SIGNAL] =
g_signal_new ("on-ice-candidate", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_STRING);
/**
* GstWebRTCBin::on-new-transceiver:
* @object: the #webrtcbin
* @candidate: the new #GstWebRTCRTPTransceiver
*/
gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL] =
g_signal_new ("on-new-transceiver", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_WEBRTC_RTP_TRANSCEIVER);
/**
* GstWebRTCBin::on-data-channel:
* @object: the #GstWebRTCBin
* @channel: the new `GstWebRTCDataChannel`
*/
gst_webrtc_bin_signals[ON_DATA_CHANNEL_SIGNAL] =
g_signal_new ("on-data-channel", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL,
G_TYPE_NONE, 1, GST_TYPE_WEBRTC_DATA_CHANNEL);
/**
* GstWebRTCBin::prepare-data-channel:
* @object: the #GstWebRTCBin
* @channel: the new `GstWebRTCDataChannel`
* @is_local: Whether this channel is local or remote
*
* Allows data-channel consumers to configure signal handlers on a newly
* created data-channel, before any data or state change has been notified.
*
* Since: 1.22
*/
gst_webrtc_bin_signals[PREPARE_DATA_CHANNEL_SIGNAL] =
g_signal_new ("prepare-data-channel", G_TYPE_FROM_CLASS (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
* @direction: the direction of the new transceiver
* @caps: (nullable): the codec preferences for this transceiver
*
* Returns: the new #GstWebRTCRTPTransceiver
*/
gst_webrtc_bin_signals[ADD_TRANSCEIVER_SIGNAL] =
g_signal_new_class_handler ("add-transceiver", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_add_transceiver), NULL, NULL,
NULL, GST_TYPE_WEBRTC_RTP_TRANSCEIVER, 2,
GST_TYPE_WEBRTC_RTP_TRANSCEIVER_DIRECTION, GST_TYPE_CAPS);
/**
* GstWebRTCBin::get-transceivers:
* @object: the #webrtcbin
*
* Returns: a #GArray of #GstWebRTCRTPTransceivers
*/
gst_webrtc_bin_signals[GET_TRANSCEIVERS_SIGNAL] =
g_signal_new_class_handler ("get-transceivers", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_get_transceivers), NULL, NULL, NULL,
G_TYPE_ARRAY, 0);
/**
* GstWebRTCBin::get-transceiver:
* @object: the #GstWebRTCBin
* @idx: The index of the transceiver
*
* Returns: (transfer full): the #GstWebRTCRTPTransceiver, or %NULL
* Since: 1.16
*/
gst_webrtc_bin_signals[GET_TRANSCEIVER_SIGNAL] =
g_signal_new_class_handler ("get-transceiver", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_get_transceiver), NULL, NULL, NULL,
GST_TYPE_WEBRTC_RTP_TRANSCEIVER, 1, G_TYPE_INT);
/**
* GstWebRTCBin::add-turn-server:
* @object: the #GstWebRTCBin
* @uri: The uri of the server of the form turn(s)://username:password@host:port
*
* Add a turn server to obtain ICE candidates from
*/
gst_webrtc_bin_signals[ADD_TURN_SERVER_SIGNAL] =
g_signal_new_class_handler ("add-turn-server", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_add_turn_server), NULL, NULL, NULL,
G_TYPE_BOOLEAN, 1, G_TYPE_STRING);
/**
* GstWebRTCBin::create-data-channel:
* @object: the #GstWebRTCBin
* @label: the label for the data channel
* @options: a #GstStructure of options for creating the data channel
*
* The options dictionary is the same format as the RTCDataChannelInit
* members outlined https://www.w3.org/TR/webrtc/#dom-rtcdatachannelinit and
* and reproduced below
*
* ordered G_TYPE_BOOLEAN Whether the channal will send data with guaranteed ordering
* max-packet-lifetime G_TYPE_INT The time in milliseconds to attempt transmitting unacknowledged data. -1 for unset
* max-retransmits G_TYPE_INT The number of times data will be attempted to be transmitted without acknowledgement before dropping
* protocol G_TYPE_STRING The subprotocol used by this channel
* negotiated G_TYPE_BOOLEAN Whether the created data channel should not perform in-band chnanel announcement. If %TRUE, then application must negotiate the channel itself and create the corresponding channel on the peer with the same id.
* id G_TYPE_INT Override the default identifier selection of this channel
* priority GST_TYPE_WEBRTC_PRIORITY_TYPE The priority to use for this channel
*
* Returns: (transfer full): a new data channel object
*/
gst_webrtc_bin_signals[CREATE_DATA_CHANNEL_SIGNAL] =
g_signal_new_class_handler ("create-data-channel",
G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_bin_create_data_channel), NULL, NULL,
NULL, GST_TYPE_WEBRTC_DATA_CHANNEL, 2, G_TYPE_STRING, GST_TYPE_STRUCTURE);
gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_PAD, 0);
gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_SINK_PAD, 0);
gst_type_mark_as_plugin_api (GST_TYPE_WEBRTC_BIN_SRC_PAD, 0);
}
static void
_unparent_and_unref (GObject * object)
{
GstObject *obj = GST_OBJECT (object);
GST_OBJECT_PARENT (obj) = NULL;
gst_object_unref (obj);
}
static void
_transport_free (GObject * object)
{
TransportStream *stream = (TransportStream *) object;
GstWebRTCBin *webrtc;
webrtc = GST_WEBRTC_BIN (GST_OBJECT_PARENT (stream));
if (stream->transport) {
g_signal_handlers_disconnect_by_data (stream->transport->transport, webrtc);
g_signal_handlers_disconnect_by_data (stream->transport, webrtc);
}
gst_object_unref (object);
}
static void
gst_webrtc_bin_init (GstWebRTCBin * webrtc)
{
/* Set SINK/SRC flags as webrtcbin can act as one depending on the
* SDP later. Without setting this here already, surrounding bins might not
* notice this and the pipeline configuration might become inconsistent,
* e.g. with regards to latency.
* See: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/737
*/
gst_bin_set_suppressed_flags (GST_BIN_CAST (webrtc),
GST_ELEMENT_FLAG_SINK | GST_ELEMENT_FLAG_SOURCE);
GST_OBJECT_FLAG_SET (webrtc, GST_ELEMENT_FLAG_SINK | GST_ELEMENT_FLAG_SOURCE);
webrtc->priv = gst_webrtc_bin_get_instance_private (webrtc);
g_mutex_init (PC_GET_LOCK (webrtc));
g_cond_init (PC_GET_COND (webrtc));
g_mutex_init (ICE_GET_LOCK (webrtc));
g_mutex_init (DC_GET_LOCK (webrtc));
webrtc->rtpbin = _create_rtpbin (webrtc);
gst_bin_add (GST_BIN (webrtc), webrtc->rtpbin);
webrtc->priv->transceivers =
g_ptr_array_new_with_free_func ((GDestroyNotify) _unparent_and_unref);
webrtc->priv->transports =
g_ptr_array_new_with_free_func ((GDestroyNotify) _transport_free);
webrtc->priv->data_channels =
g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref);
webrtc->priv->pending_data_channels =
g_ptr_array_new_with_free_func ((GDestroyNotify) gst_object_unref);
webrtc->priv->ice_stream_map =
g_array_new (FALSE, TRUE, sizeof (IceStreamItem));
webrtc->priv->pending_remote_ice_candidates =
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem));
g_array_set_clear_func (webrtc->priv->pending_remote_ice_candidates,
(GDestroyNotify) _clear_ice_candidate_item);
webrtc->priv->pending_local_ice_candidates =
g_array_new (FALSE, TRUE, sizeof (IceCandidateItem));
g_array_set_clear_func (webrtc->priv->pending_local_ice_candidates,
(GDestroyNotify) _clear_ice_candidate_item);
/* we start off closed until we move to READY */
webrtc->priv->is_closed = TRUE;
webrtc->priv->jb_latency = DEFAULT_JB_LATENCY;
}
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