mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-27 04:01:08 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/subprojects/gst-plugins-bad/ext/webrtc/gstwebrtcbin.c
Tim-Philipp Müller 99ee8af782 webrtc: fix g-i annotations for allow-none 
'allow none' doesn't exist, and 'allow-none' is now deprecated.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4146>
2023-03-10 13:09:25 +00:00

9094 lines
291 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) 2017 Matthew Waters <matthew@centricular.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "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"
/**
* 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;
}
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;
}
/* 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 void
_update_ice_gathering_state (GstWebRTCBin * webrtc)
{
gst_webrtc_bin_enqueue_task (webrtc, _update_ice_gathering_state_task, NULL,
NULL, 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_ice_transport_notify_gathering_state (GstWebRTCICETransport * transport,
GParamSpec * pspec, GstWebRTCBin * webrtc)
{
_update_ice_gathering_state (webrtc);
}
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 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_ptr_array_add (webrtc->priv->transceivers, trans);
gst_object_unref (sender);
gst_object_unref (receiver);
g_signal_emit (webrtc, gst_webrtc_bin_signals[ON_NEW_TRANSCEIVER_SIGNAL],
0, trans);
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));
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;
DC_LOCK (webrtc);
found = g_ptr_array_remove (webrtc->priv->pending_data_channels, channel)
|| g_ptr_array_remove (webrtc->priv->data_channels, channel);
if (found == FALSE) {
GST_FIXME_OBJECT (webrtc, "Received close for unknown data channel");
}
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;
channel->webrtcbin = 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);
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 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 (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;
}
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);
}
_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 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,
webrtc->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 (!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)
_add_ice_candidate_to_sdp (webrtc, webrtc->current_local_description->sdp,
item->mlineindex, cand);
if (webrtc->pending_local_description)
_add_ice_candidate_to_sdp (webrtc, webrtc->pending_local_description->sdp,
item->mlineindex, cand);
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);
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);
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);
ret->webrtcbin = webrtc;
g_ptr_array_add (webrtc->priv->data_channels, ret);
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;
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 = 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);
_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);
_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]
*
* 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]",
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
*
* 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;
}
Reference in a new issue View git blame Copy permalink