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gstreamer/ext/webrtc/webrtcdatachannel.c
Mathieu Duponchelle b42d98ca19 webrtcdatachannel: inherit directly from GObject 
There's no reason for it to inherit from GstObject apart from
locking, which is easily replaced, and inheriting from
GInitiallyUnowned made introspection awkward and needlessly
complicated.
2019-07-16 21:35:47 +00:00

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/* GStreamer
* Copyright (C) 2018 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.
*/
/**
* SECTION:gstwebrtc-datachannel
* @short_description: RTCDataChannel object
* @title: GstWebRTCDataChannel
* @see_also: #GstWebRTCRTPTransceiver
*
* <ulink url="http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport">http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport</ulink>
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include "webrtcdatachannel.h"
#include <gst/app/gstappsink.h>
#include <gst/app/gstappsrc.h>
#include <gst/base/gstbytereader.h>
#include <gst/base/gstbytewriter.h>
#include <gst/sctp/sctpreceivemeta.h>
#include <gst/sctp/sctpsendmeta.h>
#include "gstwebrtcbin.h"
#include "utils.h"
#define GST_CAT_DEFAULT gst_webrtc_data_channel_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define gst_webrtc_data_channel_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstWebRTCDataChannel, gst_webrtc_data_channel,
G_TYPE_OBJECT, GST_DEBUG_CATEGORY_INIT (gst_webrtc_data_channel_debug,
"webrtcdatachannel", 0, "webrtcdatachannel"););
#define CHANNEL_LOCK(channel) g_mutex_lock(&channel->lock)
#define CHANNEL_UNLOCK(channel) g_mutex_unlock(&channel->lock)
enum
{
SIGNAL_0,
SIGNAL_ON_OPEN,
SIGNAL_ON_CLOSE,
SIGNAL_ON_ERROR,
SIGNAL_ON_MESSAGE_DATA,
SIGNAL_ON_MESSAGE_STRING,
SIGNAL_ON_BUFFERED_AMOUNT_LOW,
SIGNAL_SEND_DATA,
SIGNAL_SEND_STRING,
SIGNAL_CLOSE,
LAST_SIGNAL,
};
enum
{
PROP_0,
PROP_LABEL,
PROP_ORDERED,
PROP_MAX_PACKET_LIFETIME,
PROP_MAX_RETRANSMITS,
PROP_PROTOCOL,
PROP_NEGOTIATED,
PROP_ID,
PROP_PRIORITY,
PROP_READY_STATE,
PROP_BUFFERED_AMOUNT,
PROP_BUFFERED_AMOUNT_LOW_THRESHOLD,
};
static guint gst_webrtc_data_channel_signals[LAST_SIGNAL] = { 0 };
typedef enum
{
DATA_CHANNEL_PPID_WEBRTC_CONTROL = 50,
DATA_CHANNEL_PPID_WEBRTC_STRING = 51,
DATA_CHANNEL_PPID_WEBRTC_BINARY_PARTIAL = 52, /* deprecated */
DATA_CHANNEL_PPID_WEBRTC_BINARY = 53,
DATA_CHANNEL_PPID_WEBRTC_STRING_PARTIAL = 54, /* deprecated */
DATA_CHANNEL_PPID_WEBRTC_BINARY_EMPTY = 56,
DATA_CHANNEL_PPID_WEBRTC_STRING_EMPTY = 57,
} DataChannelPPID;
typedef enum
{
CHANNEL_TYPE_RELIABLE = 0x00,
CHANNEL_TYPE_RELIABLE_UNORDERED = 0x80,
CHANNEL_TYPE_PARTIAL_RELIABLE_REXMIT = 0x01,
CHANNEL_TYPE_PARTIAL_RELIABLE_REXMIT_UNORDERED = 0x81,
CHANNEL_TYPE_PARTIAL_RELIABLE_TIMED = 0x02,
CHANNEL_TYPE_PARTIAL_RELIABLE_TIMED_UNORDERED = 0x82,
} DataChannelReliabilityType;
typedef enum
{
CHANNEL_MESSAGE_ACK = 0x02,
CHANNEL_MESSAGE_OPEN = 0x03,
} DataChannelMessage;
static guint16
priority_type_to_uint (GstWebRTCPriorityType pri)
{
switch (pri) {
case GST_WEBRTC_PRIORITY_TYPE_VERY_LOW:
return 64;
case GST_WEBRTC_PRIORITY_TYPE_LOW:
return 192;
case GST_WEBRTC_PRIORITY_TYPE_MEDIUM:
return 384;
case GST_WEBRTC_PRIORITY_TYPE_HIGH:
return 768;
}
g_assert_not_reached ();
return 0;
}
static GstWebRTCPriorityType
priority_uint_to_type (guint16 val)
{
if (val <= 128)
return GST_WEBRTC_PRIORITY_TYPE_VERY_LOW;
if (val <= 256)
return GST_WEBRTC_PRIORITY_TYPE_LOW;
if (val <= 512)
return GST_WEBRTC_PRIORITY_TYPE_MEDIUM;
return GST_WEBRTC_PRIORITY_TYPE_HIGH;
}
static GstBuffer *
construct_open_packet (GstWebRTCDataChannel * channel)
{
GstByteWriter w;
gsize label_len = strlen (channel->label);
gsize proto_len = strlen (channel->protocol);
gsize size = 12 + label_len + proto_len;
DataChannelReliabilityType reliability = 0;
guint32 reliability_param = 0;
guint16 priority;
GstBuffer *buf;
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Message Type | Channel Type | Priority |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Reliability Parameter |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Label Length | Protocol Length |
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* \ /
* | Label |
* / \
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* \ /
* | Protocol |
* / \
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
*/
gst_byte_writer_init_with_size (&w, size, FALSE);
if (!gst_byte_writer_put_uint8 (&w, (guint8) CHANNEL_MESSAGE_OPEN))
g_return_val_if_reached (NULL);
if (!channel->ordered)
reliability |= 0x80;
if (channel->max_retransmits != -1) {
reliability |= 0x01;
reliability_param = channel->max_retransmits;
}
if (channel->max_packet_lifetime != -1) {
reliability |= 0x02;
reliability_param = channel->max_packet_lifetime;
}
priority = priority_type_to_uint (channel->priority);
if (!gst_byte_writer_put_uint8 (&w, (guint8) reliability))
g_return_val_if_reached (NULL);
if (!gst_byte_writer_put_uint16_be (&w, (guint16) priority))
g_return_val_if_reached (NULL);
if (!gst_byte_writer_put_uint32_be (&w, (guint32) reliability_param))
g_return_val_if_reached (NULL);
if (!gst_byte_writer_put_uint16_be (&w, (guint16) label_len))
g_return_val_if_reached (NULL);
if (!gst_byte_writer_put_uint16_be (&w, (guint16) proto_len))
g_return_val_if_reached (NULL);
if (!gst_byte_writer_put_data (&w, (guint8 *) channel->label, label_len))
g_return_val_if_reached (NULL);
if (!gst_byte_writer_put_data (&w, (guint8 *) channel->protocol, proto_len))
g_return_val_if_reached (NULL);
buf = gst_byte_writer_reset_and_get_buffer (&w);
/* send reliable and ordered */
gst_sctp_buffer_add_send_meta (buf, DATA_CHANNEL_PPID_WEBRTC_CONTROL, TRUE,
GST_SCTP_SEND_META_PARTIAL_RELIABILITY_NONE, 0);
return buf;
}
static GstBuffer *
construct_ack_packet (GstWebRTCDataChannel * channel)
{
GstByteWriter w;
GstBuffer *buf;
/*
* 0 1 2 3
* 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
* +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
* | Message Type |
* +-+-+-+-+-+-+-+-+
*/
gst_byte_writer_init_with_size (&w, 1, FALSE);
if (!gst_byte_writer_put_uint8 (&w, (guint8) CHANNEL_MESSAGE_ACK))
g_return_val_if_reached (NULL);
buf = gst_byte_writer_reset_and_get_buffer (&w);
/* send reliable and ordered */
gst_sctp_buffer_add_send_meta (buf, DATA_CHANNEL_PPID_WEBRTC_CONTROL, TRUE,
GST_SCTP_SEND_META_PARTIAL_RELIABILITY_NONE, 0);
return buf;
}
typedef void (*ChannelTask) (GstWebRTCDataChannel * channel,
gpointer user_data);
struct task
{
GstWebRTCDataChannel *channel;
ChannelTask func;
gpointer user_data;
GDestroyNotify notify;
};
static void
_execute_task (GstWebRTCBin * webrtc, struct task *task)
{
if (task->func)
task->func (task->channel, task->user_data);
}
static void
_free_task (struct task *task)
{
gst_object_unref (task->channel);
if (task->notify)
task->notify (task->user_data);
g_free (task);
}
static void
_channel_enqueue_task (GstWebRTCDataChannel * channel, ChannelTask func,
gpointer user_data, GDestroyNotify notify)
{
struct task *task = g_new0 (struct task, 1);
task->channel = gst_object_ref (channel);
task->func = func;
task->user_data = user_data;
task->notify = notify;
gst_webrtc_bin_enqueue_task (channel->webrtcbin,
(GstWebRTCBinFunc) _execute_task, task, (GDestroyNotify) _free_task);
}
static void
_channel_store_error (GstWebRTCDataChannel * channel, GError * error)
{
CHANNEL_LOCK (channel);
if (error) {
GST_WARNING_OBJECT (channel, "Error: %s",
error ? error->message : "Unknown");
if (!channel->stored_error)
channel->stored_error = error;
else
g_clear_error (&error);
}
CHANNEL_UNLOCK (channel);
}
static void
_maybe_emit_on_error (GstWebRTCDataChannel * channel, GError * error)
{
if (error) {
GST_WARNING_OBJECT (channel, "error thrown");
g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_ERROR], 0,
error);
}
}
static void
_emit_on_open (GstWebRTCDataChannel * channel, gpointer user_data)
{
CHANNEL_LOCK (channel);
if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING ||
channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
CHANNEL_UNLOCK (channel);
return;
}
if (channel->ready_state != GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
channel->ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_OPEN;
CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (channel), "ready-state");
GST_INFO_OBJECT (channel, "We are open and ready for data!");
g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_OPEN], 0,
NULL);
} else {
CHANNEL_UNLOCK (channel);
}
}
static void
_transport_closed_unlocked (GstWebRTCDataChannel * channel)
{
GError *error;
if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED)
return;
channel->ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED;
error = channel->stored_error;
channel->stored_error = NULL;
CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (channel), "ready-state");
GST_INFO_OBJECT (channel, "We are closed for data");
_maybe_emit_on_error (channel, error);
g_signal_emit (channel, gst_webrtc_data_channel_signals[SIGNAL_ON_CLOSE], 0,
NULL);
CHANNEL_LOCK (channel);
}
static void
_transport_closed (GstWebRTCDataChannel * channel, gpointer user_data)
{
CHANNEL_LOCK (channel);
_transport_closed_unlocked (channel);
CHANNEL_UNLOCK (channel);
}
static void
_close_sctp_stream (GstWebRTCDataChannel * channel, gpointer user_data)
{
GstPad *pad, *peer;
pad = gst_element_get_static_pad (channel->appsrc, "src");
peer = gst_pad_get_peer (pad);
gst_object_unref (pad);
if (peer) {
GstElement *sctpenc = gst_pad_get_parent_element (peer);
if (sctpenc) {
gst_element_release_request_pad (sctpenc, peer);
gst_object_unref (sctpenc);
}
gst_object_unref (peer);
}
_transport_closed (channel, NULL);
}
static void
_close_procedure (GstWebRTCDataChannel * channel, gpointer user_data)
{
/* https://www.w3.org/TR/webrtc/#data-transport-closing-procedure */
CHANNEL_LOCK (channel);
if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED
|| channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING) {
CHANNEL_UNLOCK (channel);
return;
}
channel->ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING;
CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (channel), "ready-state");
CHANNEL_LOCK (channel);
if (channel->buffered_amount <= 0) {
_channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream,
NULL, NULL);
}
CHANNEL_UNLOCK (channel);
}
static void
_on_sctp_reset_stream (GstWebRTCSCTPTransport * sctp, guint stream_id,
GstWebRTCDataChannel * channel)
{
if (channel->id == stream_id)
_channel_enqueue_task (channel, (ChannelTask) _transport_closed,
GUINT_TO_POINTER (stream_id), NULL);
}
static void
gst_webrtc_data_channel_close (GstWebRTCDataChannel * channel)
{
_close_procedure (channel, NULL);
}
static GstFlowReturn
_parse_control_packet (GstWebRTCDataChannel * channel, guint8 * data,
gsize size, GError ** error)
{
GstByteReader r;
guint8 message_type;
if (!data)
g_return_val_if_reached (GST_FLOW_ERROR);
if (size < 1)
g_return_val_if_reached (GST_FLOW_ERROR);
gst_byte_reader_init (&r, data, size);
if (!gst_byte_reader_get_uint8 (&r, &message_type))
g_return_val_if_reached (GST_FLOW_ERROR);
if (message_type == CHANNEL_MESSAGE_ACK) {
/* all good */
GST_INFO_OBJECT (channel, "Received channel ack");
return GST_FLOW_OK;
} else if (message_type == CHANNEL_MESSAGE_OPEN) {
guint8 reliability;
guint32 reliability_param;
guint16 priority, label_len, proto_len;
const guint8 *src;
gchar *label, *proto;
GstBuffer *buffer;
GstFlowReturn ret;
GST_INFO_OBJECT (channel, "Received channel open");
if (channel->negotiated) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Data channel was signalled as negotiated already");
g_return_val_if_reached (GST_FLOW_ERROR);
}
if (channel->opened)
return GST_FLOW_OK;
if (!gst_byte_reader_get_uint8 (&r, &reliability))
goto parse_error;
if (!gst_byte_reader_get_uint16_be (&r, &priority))
goto parse_error;
if (!gst_byte_reader_get_uint32_be (&r, &reliability_param))
goto parse_error;
if (!gst_byte_reader_get_uint16_be (&r, &label_len))
goto parse_error;
if (!gst_byte_reader_get_uint16_be (&r, &proto_len))
goto parse_error;
label = g_new0 (gchar, (gsize) label_len + 1);
proto = g_new0 (gchar, (gsize) proto_len + 1);
if (!gst_byte_reader_get_data (&r, label_len, &src))
goto parse_error;
memcpy (label, src, label_len);
label[label_len] = '\0';
if (!gst_byte_reader_get_data (&r, proto_len, &src))
goto parse_error;
memcpy (proto, src, proto_len);
proto[proto_len] = '\0';
channel->label = label;
channel->protocol = proto;
channel->priority = priority_uint_to_type (priority);
channel->ordered = !(reliability & 0x80);
if (reliability & 0x01) {
channel->max_retransmits = reliability_param;
channel->max_packet_lifetime = -1;
} else if (reliability & 0x02) {
channel->max_retransmits = -1;
channel->max_packet_lifetime = reliability_param;
} else {
channel->max_retransmits = -1;
channel->max_packet_lifetime = -1;
}
channel->opened = TRUE;
GST_INFO_OBJECT (channel, "Received channel open for SCTP stream %i "
"label %s protocol %s ordered %s", channel->id, channel->label,
channel->protocol, channel->ordered ? "true" : "false");
_channel_enqueue_task (channel, (ChannelTask) _emit_on_open, NULL, NULL);
GST_INFO_OBJECT (channel, "Sending channel ack");
buffer = construct_ack_packet (channel);
CHANNEL_LOCK (channel);
channel->buffered_amount += gst_buffer_get_size (buffer);
CHANNEL_UNLOCK (channel);
ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer);
if (ret != GST_FLOW_OK) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Could not send ack packet");
}
return ret;
} else {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Unknown message type in control protocol");
return GST_FLOW_ERROR;
}
parse_error:
{
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "Failed to parse packet");
g_return_val_if_reached (GST_FLOW_ERROR);
}
}
static void
on_sink_eos (GstAppSink * sink, gpointer user_data)
{
}
struct map_info
{
GstBuffer *buffer;
GstMapInfo map_info;
};
static void
buffer_unmap_and_unref (struct map_info *info)
{
gst_buffer_unmap (info->buffer, &info->map_info);
gst_buffer_unref (info->buffer);
g_free (info);
}
static void
_emit_have_data (GstWebRTCDataChannel * channel, GBytes * data)
{
GST_LOG_OBJECT (channel, "Have data %p", data);
g_signal_emit (channel,
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_DATA], 0, data);
}
static void
_emit_have_string (GstWebRTCDataChannel * channel, gchar * str)
{
GST_LOG_OBJECT (channel, "Have string %p", str);
g_signal_emit (channel,
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_STRING], 0, str);
}
static GstFlowReturn
_data_channel_have_sample (GstWebRTCDataChannel * channel, GstSample * sample,
GError ** error)
{
GstSctpReceiveMeta *receive;
GstBuffer *buffer;
GstFlowReturn ret = GST_FLOW_OK;
GST_LOG_OBJECT (channel, "Received sample %" GST_PTR_FORMAT, sample);
g_return_val_if_fail (channel->sctp_transport != NULL, GST_FLOW_ERROR);
buffer = gst_sample_get_buffer (sample);
if (!buffer) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "No buffer to handle");
return GST_FLOW_ERROR;
}
receive = gst_sctp_buffer_get_receive_meta (buffer);
if (!receive) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"No SCTP Receive meta on the buffer");
return GST_FLOW_ERROR;
}
switch (receive->ppid) {
case DATA_CHANNEL_PPID_WEBRTC_CONTROL:{
GstMapInfo info = GST_MAP_INFO_INIT;
if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Failed to map received buffer");
ret = GST_FLOW_ERROR;
} else {
ret = _parse_control_packet (channel, info.data, info.size, error);
}
break;
}
case DATA_CHANNEL_PPID_WEBRTC_STRING:
case DATA_CHANNEL_PPID_WEBRTC_STRING_PARTIAL:{
GstMapInfo info = GST_MAP_INFO_INIT;
if (!gst_buffer_map (buffer, &info, GST_MAP_READ)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Failed to map received buffer");
ret = GST_FLOW_ERROR;
} else {
gchar *str = g_strndup ((gchar *) info.data, info.size);
_channel_enqueue_task (channel, (ChannelTask) _emit_have_string, str,
g_free);
}
break;
}
case DATA_CHANNEL_PPID_WEBRTC_BINARY:
case DATA_CHANNEL_PPID_WEBRTC_BINARY_PARTIAL:{
struct map_info *info = g_new0 (struct map_info, 1);
if (!gst_buffer_map (buffer, &info->map_info, GST_MAP_READ)) {
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Failed to map received buffer");
ret = GST_FLOW_ERROR;
} else {
GBytes *data = g_bytes_new_with_free_func (info->map_info.data,
info->map_info.size, (GDestroyNotify) buffer_unmap_and_unref, info);
info->buffer = gst_buffer_ref (buffer);
_channel_enqueue_task (channel, (ChannelTask) _emit_have_data, data,
(GDestroyNotify) g_bytes_unref);
}
break;
}
case DATA_CHANNEL_PPID_WEBRTC_BINARY_EMPTY:
_channel_enqueue_task (channel, (ChannelTask) _emit_have_data, NULL,
NULL);
break;
case DATA_CHANNEL_PPID_WEBRTC_STRING_EMPTY:
_channel_enqueue_task (channel, (ChannelTask) _emit_have_string, NULL,
NULL);
break;
default:
g_set_error (error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Unknown SCTP PPID %u received", receive->ppid);
ret = GST_FLOW_ERROR;
break;
}
return ret;
}
static GstFlowReturn
on_sink_preroll (GstAppSink * sink, gpointer user_data)
{
GstWebRTCDataChannel *channel = user_data;
GstSample *sample = gst_app_sink_pull_preroll (sink);
GstFlowReturn ret;
if (sample) {
/* This sample also seems to be provided by the sample callback
ret = _data_channel_have_sample (channel, sample); */
ret = GST_FLOW_OK;
gst_sample_unref (sample);
} else if (gst_app_sink_is_eos (sink)) {
ret = GST_FLOW_EOS;
} else {
ret = GST_FLOW_ERROR;
}
if (ret != GST_FLOW_OK) {
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}
return ret;
}
static GstFlowReturn
on_sink_sample (GstAppSink * sink, gpointer user_data)
{
GstWebRTCDataChannel *channel = user_data;
GstSample *sample = gst_app_sink_pull_sample (sink);
GstFlowReturn ret;
GError *error = NULL;
if (sample) {
ret = _data_channel_have_sample (channel, sample, &error);
gst_sample_unref (sample);
} else if (gst_app_sink_is_eos (sink)) {
ret = GST_FLOW_EOS;
} else {
ret = GST_FLOW_ERROR;
}
if (error)
_channel_store_error (channel, error);
if (ret != GST_FLOW_OK) {
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}
return ret;
}
static GstAppSinkCallbacks sink_callbacks = {
on_sink_eos,
on_sink_preroll,
on_sink_sample,
};
void
gst_webrtc_data_channel_start_negotiation (GstWebRTCDataChannel * channel)
{
GstBuffer *buffer;
g_return_if_fail (!channel->negotiated);
g_return_if_fail (channel->id != -1);
g_return_if_fail (channel->sctp_transport != NULL);
buffer = construct_open_packet (channel);
GST_INFO_OBJECT (channel, "Sending channel open for SCTP stream %i "
"label %s protocol %s ordered %s", channel->id, channel->label,
channel->protocol, channel->ordered ? "true" : "false");
CHANNEL_LOCK (channel);
channel->buffered_amount += gst_buffer_get_size (buffer);
CHANNEL_UNLOCK (channel);
if (gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc),
buffer) == GST_FLOW_OK) {
channel->opened = TRUE;
_channel_enqueue_task (channel, (ChannelTask) _emit_on_open, NULL, NULL);
} else {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Failed to send DCEP open packet");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}
}
static void
_get_sctp_reliability (GstWebRTCDataChannel * channel,
GstSctpSendMetaPartiallyReliability * reliability, guint * rel_param)
{
if (channel->max_retransmits != -1) {
*reliability = GST_SCTP_SEND_META_PARTIAL_RELIABILITY_RTX;
*rel_param = channel->max_retransmits;
} else if (channel->max_packet_lifetime != -1) {
*reliability = GST_SCTP_SEND_META_PARTIAL_RELIABILITY_TTL;
*rel_param = channel->max_packet_lifetime;
} else {
*reliability = GST_SCTP_SEND_META_PARTIAL_RELIABILITY_NONE;
*rel_param = 0;
}
}
static gboolean
_is_within_max_message_size (GstWebRTCDataChannel * channel, gsize size)
{
return size <= channel->sctp_transport->max_message_size;
}
static void
gst_webrtc_data_channel_send_data (GstWebRTCDataChannel * channel,
GBytes * bytes)
{
GstSctpSendMetaPartiallyReliability reliability;
guint rel_param;
guint32 ppid;
GstBuffer *buffer;
GstFlowReturn ret;
if (!bytes) {
buffer = gst_buffer_new ();
ppid = DATA_CHANNEL_PPID_WEBRTC_BINARY_EMPTY;
} else {
gsize size;
guint8 *data;
data = (guint8 *) g_bytes_get_data (bytes, &size);
g_return_if_fail (data != NULL);
if (!_is_within_max_message_size (channel, size)) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Requested to send data that is too large");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL,
NULL);
return;
}
buffer = gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY, data, size,
0, size, g_bytes_ref (bytes), (GDestroyNotify) g_bytes_unref);
ppid = DATA_CHANNEL_PPID_WEBRTC_BINARY;
}
_get_sctp_reliability (channel, &reliability, &rel_param);
gst_sctp_buffer_add_send_meta (buffer, ppid, channel->ordered, reliability,
rel_param);
GST_LOG_OBJECT (channel, "Sending data using buffer %" GST_PTR_FORMAT,
buffer);
CHANNEL_LOCK (channel);
channel->buffered_amount += gst_buffer_get_size (buffer);
CHANNEL_UNLOCK (channel);
ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer);
if (ret != GST_FLOW_OK) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "Failed to send data");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}
}
static void
gst_webrtc_data_channel_send_string (GstWebRTCDataChannel * channel,
gchar * str)
{
GstSctpSendMetaPartiallyReliability reliability;
guint rel_param;
guint32 ppid;
GstBuffer *buffer;
GstFlowReturn ret;
if (!channel->negotiated)
g_return_if_fail (channel->opened);
g_return_if_fail (channel->sctp_transport != NULL);
if (!str) {
buffer = gst_buffer_new ();
ppid = DATA_CHANNEL_PPID_WEBRTC_STRING_EMPTY;
} else {
gsize size = strlen (str);
gchar *str_copy = g_strdup (str);
if (!_is_within_max_message_size (channel, size)) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE,
"Requested to send a string that is too large");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL,
NULL);
return;
}
buffer =
gst_buffer_new_wrapped_full (GST_MEMORY_FLAG_READONLY, str_copy,
size, 0, size, str_copy, g_free);
ppid = DATA_CHANNEL_PPID_WEBRTC_STRING;
}
_get_sctp_reliability (channel, &reliability, &rel_param);
gst_sctp_buffer_add_send_meta (buffer, ppid, channel->ordered, reliability,
rel_param);
GST_TRACE_OBJECT (channel, "Sending string using buffer %" GST_PTR_FORMAT,
buffer);
CHANNEL_LOCK (channel);
channel->buffered_amount += gst_buffer_get_size (buffer);
CHANNEL_UNLOCK (channel);
ret = gst_app_src_push_buffer (GST_APP_SRC (channel->appsrc), buffer);
if (ret != GST_FLOW_OK) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_BIN_ERROR,
GST_WEBRTC_BIN_ERROR_DATA_CHANNEL_FAILURE, "Failed to send string");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}
}
static void
_on_sctp_notify_state_unlocked (GObject * sctp_transport,
GstWebRTCDataChannel * channel)
{
GstWebRTCSCTPTransportState state;
g_object_get (sctp_transport, "state", &state, NULL);
if (state == GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED) {
if (channel->negotiated)
_channel_enqueue_task (channel, (ChannelTask) _emit_on_open, NULL, NULL);
}
}
static void
_on_sctp_notify_state (GObject * sctp_transport, GParamSpec * pspec,
GstWebRTCDataChannel * channel)
{
CHANNEL_LOCK (channel);
_on_sctp_notify_state_unlocked (sctp_transport, channel);
CHANNEL_UNLOCK (channel);
}
static void
gst_webrtc_data_channel_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
CHANNEL_LOCK (channel);
switch (prop_id) {
case PROP_LABEL:
channel->label = g_value_dup_string (value);
break;
case PROP_ORDERED:
channel->ordered = g_value_get_boolean (value);
break;
case PROP_MAX_PACKET_LIFETIME:
channel->max_packet_lifetime = g_value_get_int (value);
break;
case PROP_MAX_RETRANSMITS:
channel->max_retransmits = g_value_get_int (value);
break;
case PROP_PROTOCOL:
channel->protocol = g_value_dup_string (value);
break;
case PROP_NEGOTIATED:
channel->negotiated = g_value_get_boolean (value);
break;
case PROP_ID:
channel->id = g_value_get_int (value);
break;
case PROP_PRIORITY:
channel->priority = g_value_get_enum (value);
break;
case PROP_BUFFERED_AMOUNT_LOW_THRESHOLD:
channel->buffered_amount_low_threshold = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
CHANNEL_UNLOCK (channel);
}
static void
gst_webrtc_data_channel_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
CHANNEL_LOCK (channel);
switch (prop_id) {
case PROP_LABEL:
g_value_set_string (value, channel->label);
break;
case PROP_ORDERED:
g_value_set_boolean (value, channel->ordered);
break;
case PROP_MAX_PACKET_LIFETIME:
g_value_set_int (value, channel->max_packet_lifetime);
break;
case PROP_MAX_RETRANSMITS:
g_value_set_int (value, channel->max_retransmits);
break;
case PROP_PROTOCOL:
g_value_set_string (value, channel->protocol);
break;
case PROP_NEGOTIATED:
g_value_set_boolean (value, channel->negotiated);
break;
case PROP_ID:
g_value_set_int (value, channel->id);
break;
case PROP_PRIORITY:
g_value_set_enum (value, channel->priority);
break;
case PROP_READY_STATE:
g_value_set_enum (value, channel->ready_state);
break;
case PROP_BUFFERED_AMOUNT:
g_value_set_uint64 (value, channel->buffered_amount);
break;
case PROP_BUFFERED_AMOUNT_LOW_THRESHOLD:
g_value_set_uint64 (value, channel->buffered_amount_low_threshold);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
CHANNEL_UNLOCK (channel);
}
static void
_emit_low_threshold (GstWebRTCDataChannel * channel, gpointer user_data)
{
GST_LOG_OBJECT (channel, "Low threshold reached");
g_signal_emit (channel,
gst_webrtc_data_channel_signals[SIGNAL_ON_BUFFERED_AMOUNT_LOW], 0);
}
static GstPadProbeReturn
on_appsrc_data (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
GstWebRTCDataChannel *channel = user_data;
guint64 prev_amount;
guint64 size = 0;
if (GST_PAD_PROBE_INFO_TYPE (info) & (GST_PAD_PROBE_TYPE_BUFFER)) {
GstBuffer *buffer = GST_PAD_PROBE_INFO_BUFFER (info);
size = gst_buffer_get_size (buffer);
} else if (GST_PAD_PROBE_INFO_TYPE (info) & GST_PAD_PROBE_TYPE_BUFFER_LIST) {
GstBufferList *list = GST_PAD_PROBE_INFO_BUFFER_LIST (info);
size = gst_buffer_list_calculate_size (list);
}
if (size > 0) {
CHANNEL_LOCK (channel);
prev_amount = channel->buffered_amount;
channel->buffered_amount -= size;
if (prev_amount > channel->buffered_amount_low_threshold &&
channel->buffered_amount < channel->buffered_amount_low_threshold) {
_channel_enqueue_task (channel, (ChannelTask) _emit_low_threshold,
NULL, NULL);
}
if (channel->ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING
&& channel->buffered_amount <= 0) {
_channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream, NULL,
NULL);
}
CHANNEL_UNLOCK (channel);
}
return GST_PAD_PROBE_OK;
}
static void
gst_webrtc_data_channel_constructed (GObject * object)
{
GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
GstPad *pad;
GstCaps *caps;
caps = gst_caps_new_any ();
channel->appsrc = gst_element_factory_make ("appsrc", NULL);
gst_object_ref_sink (channel->appsrc);
pad = gst_element_get_static_pad (channel->appsrc, "src");
channel->src_probe = gst_pad_add_probe (pad, GST_PAD_PROBE_TYPE_DATA_BOTH,
(GstPadProbeCallback) on_appsrc_data, channel, NULL);
channel->appsink = gst_element_factory_make ("appsink", NULL);
gst_object_ref_sink (channel->appsink);
g_object_set (channel->appsink, "sync", FALSE, "async", FALSE, "caps", caps,
NULL);
gst_app_sink_set_callbacks (GST_APP_SINK (channel->appsink), &sink_callbacks,
channel, NULL);
gst_object_unref (pad);
gst_caps_unref (caps);
}
static void
gst_webrtc_data_channel_finalize (GObject * object)
{
GstWebRTCDataChannel *channel = GST_WEBRTC_DATA_CHANNEL (object);
if (channel->src_probe) {
GstPad *pad = gst_element_get_static_pad (channel->appsrc, "src");
gst_pad_remove_probe (pad, channel->src_probe);
gst_object_unref (pad);
channel->src_probe = 0;
}
g_free (channel->label);
channel->label = NULL;
g_free (channel->protocol);
channel->protocol = NULL;
if (channel->sctp_transport)
g_signal_handlers_disconnect_by_data (channel->sctp_transport, channel);
g_clear_object (&channel->sctp_transport);
g_clear_object (&channel->appsrc);
g_clear_object (&channel->appsink);
g_mutex_clear (&channel->lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_webrtc_data_channel_class_init (GstWebRTCDataChannelClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
gobject_class->constructed = gst_webrtc_data_channel_constructed;
gobject_class->get_property = gst_webrtc_data_channel_get_property;
gobject_class->set_property = gst_webrtc_data_channel_set_property;
gobject_class->finalize = gst_webrtc_data_channel_finalize;
g_object_class_install_property (gobject_class,
PROP_LABEL,
g_param_spec_string ("label",
"Label", "Data channel label",
NULL,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ORDERED,
g_param_spec_boolean ("ordered",
"Ordered", "Using ordered transmission mode",
FALSE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MAX_PACKET_LIFETIME,
g_param_spec_int ("max-packet-lifetime",
"Maximum Packet Lifetime",
"Maximum number of milliseconds that transmissions and "
"retransmissions may occur in unreliable mode (-1 = unset)",
-1, G_MAXUINT16, -1,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_MAX_RETRANSMITS,
g_param_spec_int ("max-retransmits",
"Maximum Retransmits",
"Maximum number of retransmissions attempted in unreliable mode",
-1, G_MAXUINT16, 0,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_PROTOCOL,
g_param_spec_string ("protocol",
"Protocol", "Data channel protocol",
"",
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_NEGOTIATED,
g_param_spec_boolean ("negotiated",
"Negotiated",
"Whether this data channel was negotiated by the application", FALSE,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_ID,
g_param_spec_int ("id",
"ID",
"ID negotiated by this data channel (-1 = unset)",
-1, G_MAXUINT16, -1,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_PRIORITY,
g_param_spec_enum ("priority",
"Priority",
"The priority of data sent using this data channel",
GST_TYPE_WEBRTC_PRIORITY_TYPE,
GST_WEBRTC_PRIORITY_TYPE_LOW,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_READY_STATE,
g_param_spec_enum ("ready-state",
"Ready State",
"The Ready state of this data channel",
GST_TYPE_WEBRTC_DATA_CHANNEL_STATE,
GST_WEBRTC_DATA_CHANNEL_STATE_NEW,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_BUFFERED_AMOUNT,
g_param_spec_uint64 ("buffered-amount",
"Buffered Amount",
"The amount of data in bytes currently buffered",
0, G_MAXUINT64, 0, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_BUFFERED_AMOUNT_LOW_THRESHOLD,
g_param_spec_uint64 ("buffered-amount-low-threshold",
"Buffered Amount Low Threshold",
"The threshold at which the buffered amount is considered low and "
"the buffered-amount-low signal is emitted",
0, G_MAXUINT64, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstWebRTCDataChannel::on-open:
* @object: the #GstWebRTCDataChannel
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_OPEN] =
g_signal_new ("on-open", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 0);
/**
* GstWebRTCDataChannel::on-close:
* @object: the #GstWebRTCDataChannel
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_CLOSE] =
g_signal_new ("on-close", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 0);
/**
* GstWebRTCDataChannel::on-error:
* @object: the #GstWebRTCDataChannel
* @error: the #GError thrown
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_ERROR] =
g_signal_new ("on-error", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, G_TYPE_ERROR);
/**
* GstWebRTCDataChannel::on-message-data:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): a #GBytes of the data received
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_DATA] =
g_signal_new ("on-message-data", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, G_TYPE_BYTES);
/**
* GstWebRTCDataChannel::on-message-string:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): the data received as a string
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_MESSAGE_STRING] =
g_signal_new ("on-message-string", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 1, G_TYPE_STRING);
/**
* GstWebRTCDataChannel::on-buffered-amount-low:
* @object: the #GstWebRTCDataChannel
*/
gst_webrtc_data_channel_signals[SIGNAL_ON_BUFFERED_AMOUNT_LOW] =
g_signal_new ("on-buffered-amount-low", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, 0, NULL, NULL, g_cclosure_marshal_generic,
G_TYPE_NONE, 0);
/**
* GstWebRTCDataChannel::send-data:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): a #GBytes with the data
*/
gst_webrtc_data_channel_signals[SIGNAL_SEND_DATA] =
g_signal_new_class_handler ("send-data", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_data_channel_send_data), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_BYTES);
/**
* GstWebRTCDataChannel::send-string:
* @object: the #GstWebRTCDataChannel
* @data: (nullable): the data to send as a string
*/
gst_webrtc_data_channel_signals[SIGNAL_SEND_STRING] =
g_signal_new_class_handler ("send-string", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_data_channel_send_string), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_STRING);
/**
* GstWebRTCDataChannel::close:
* @object: the #GstWebRTCDataChannel
*
* Close the data channel
*/
gst_webrtc_data_channel_signals[SIGNAL_CLOSE] =
g_signal_new_class_handler ("close", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_CALLBACK (gst_webrtc_data_channel_close), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 0);
}
static void
gst_webrtc_data_channel_init (GstWebRTCDataChannel * channel)
{
g_mutex_init (&channel->lock);
}
static void
_data_channel_set_sctp_transport (GstWebRTCDataChannel * channel,
GstWebRTCSCTPTransport * sctp)
{
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
g_return_if_fail (GST_IS_WEBRTC_SCTP_TRANSPORT (sctp));
CHANNEL_LOCK (channel);
if (channel->sctp_transport)
g_signal_handlers_disconnect_by_data (channel->sctp_transport, channel);
gst_object_replace ((GstObject **) & channel->sctp_transport,
GST_OBJECT (sctp));
if (sctp) {
g_signal_connect (sctp, "stream-reset", G_CALLBACK (_on_sctp_reset_stream),
channel);
g_signal_connect (sctp, "notify::state", G_CALLBACK (_on_sctp_notify_state),
channel);
_on_sctp_notify_state_unlocked (G_OBJECT (sctp), channel);
}
CHANNEL_UNLOCK (channel);
}
void
gst_webrtc_data_channel_link_to_sctp (GstWebRTCDataChannel * channel,
GstWebRTCSCTPTransport * sctp_transport)
{
if (sctp_transport && !channel->sctp_transport) {
gint id;
g_object_get (channel, "id", &id, NULL);
if (sctp_transport->association_established && id != -1) {
gchar *pad_name;
_data_channel_set_sctp_transport (channel, sctp_transport);
pad_name = g_strdup_printf ("sink_%u", id);
if (!gst_element_link_pads (channel->appsrc, "src",
channel->sctp_transport->sctpenc, pad_name))
g_warn_if_reached ();
g_free (pad_name);
}
}
}
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