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gstreamer/subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c
Philippe Normand 4254920b72 webrtc: Expose RTCError enum 
The error codes not complying with the spec are now notified with the
GST_WEBRTC_ERROR_INTERNAL_FAILURE code.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1485>
2022-01-29 14:42:22 +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
*
* <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransport>
*/
#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 webrtc_data_channel_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
#define webrtc_data_channel_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (WebRTCDataChannel, webrtc_data_channel,
GST_TYPE_WEBRTC_DATA_CHANNEL,
GST_DEBUG_CATEGORY_INIT (webrtc_data_channel_debug, "webrtcdatachannel", 0,
"webrtcdatachannel"););
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 (WebRTCDataChannel * channel)
{
GstByteWriter w;
gsize label_len = strlen (channel->parent.label);
gsize proto_len = strlen (channel->parent.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->parent.ordered)
reliability |= 0x80;
if (channel->parent.max_retransmits != -1) {
reliability |= 0x01;
reliability_param = channel->parent.max_retransmits;
}
if (channel->parent.max_packet_lifetime != -1) {
reliability |= 0x02;
reliability_param = channel->parent.max_packet_lifetime;
}
priority = priority_type_to_uint (channel->parent.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->parent.label,
label_len))
g_return_val_if_reached (NULL);
if (!gst_byte_writer_put_data (&w, (guint8 *) channel->parent.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 (WebRTCDataChannel * 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 GstStructure *
_execute_task (GstWebRTCBin * webrtc, struct task *task)
{
if (task->func)
task->func (task->channel, task->user_data);
return NULL;
}
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 (WebRTCDataChannel * 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,
NULL);
}
static void
_channel_store_error (WebRTCDataChannel * channel, GError * error)
{
GST_WEBRTC_DATA_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);
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
}
static void
_emit_on_open (WebRTCDataChannel * channel, gpointer user_data)
{
gst_webrtc_data_channel_on_open (GST_WEBRTC_DATA_CHANNEL (channel));
}
static void
_transport_closed (WebRTCDataChannel * channel)
{
GError *error;
gboolean both_sides_closed;
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
error = channel->stored_error;
channel->stored_error = NULL;
both_sides_closed =
channel->peer_closed && channel->parent.buffered_amount <= 0;
if (both_sides_closed || error) {
channel->peer_closed = FALSE;
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
if (error) {
gst_webrtc_data_channel_on_error (GST_WEBRTC_DATA_CHANNEL (channel), error);
g_clear_error (&error);
}
if (both_sides_closed || error) {
gst_webrtc_data_channel_on_close (GST_WEBRTC_DATA_CHANNEL (channel));
}
}
static void
_close_sctp_stream (WebRTCDataChannel * channel, gpointer user_data)
{
GstPad *pad, *peer;
GST_INFO_OBJECT (channel, "Closing outgoing SCTP stream %i label \"%s\"",
channel->parent.id, channel->parent.label);
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);
}
static void
_close_procedure (WebRTCDataChannel * channel, gpointer user_data)
{
/* https://www.w3.org/TR/webrtc/#data-transport-closing-procedure */
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
if (channel->parent.ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSED) {
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
return;
} else if (channel->parent.ready_state ==
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING) {
_channel_enqueue_task (channel, (ChannelTask) _transport_closed, NULL,
NULL);
} else if (channel->parent.ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_OPEN) {
channel->parent.ready_state = GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING;
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (channel), "ready-state");
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
if (channel->parent.buffered_amount <= 0) {
_channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream,
NULL, NULL);
}
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
}
static void
_on_sctp_stream_reset (WebRTCSCTPTransport * sctp, guint stream_id,
WebRTCDataChannel * channel)
{
if (channel->parent.id == stream_id) {
GST_INFO_OBJECT (channel,
"Received channel close for SCTP stream %i label \"%s\"",
channel->parent.id, channel->parent.label);
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
channel->peer_closed = TRUE;
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure,
GUINT_TO_POINTER (stream_id), NULL);
}
}
static void
webrtc_data_channel_close (GstWebRTCDataChannel * channel)
{
_close_procedure (WEBRTC_DATA_CHANNEL (channel), NULL);
}
static GstFlowReturn
_parse_control_packet (WebRTCDataChannel * channel, guint8 * data,
gsize size, GError ** error)
{
GstByteReader r;
guint8 message_type;
gchar *label = NULL;
gchar *proto = NULL;
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;
GstBuffer *buffer;
GstFlowReturn ret;
GST_INFO_OBJECT (channel, "Received channel open");
if (channel->parent.negotiated) {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_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';
g_free (channel->parent.label);
channel->parent.label = label;
g_free (channel->parent.protocol);
channel->parent.protocol = proto;
channel->parent.priority = priority_uint_to_type (priority);
channel->parent.ordered = !(reliability & 0x80);
if (reliability & 0x01) {
channel->parent.max_retransmits = reliability_param;
channel->parent.max_packet_lifetime = -1;
} else if (reliability & 0x02) {
channel->parent.max_retransmits = -1;
channel->parent.max_packet_lifetime = reliability_param;
} else {
channel->parent.max_retransmits = -1;
channel->parent.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->parent.id,
channel->parent.label, channel->parent.protocol,
channel->parent.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);
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
channel->parent.buffered_amount += gst_buffer_get_size (buffer);
GST_WEBRTC_DATA_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_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, "Could not send ack packet");
return ret;
}
return ret;
} else {
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Unknown message type in control protocol");
return GST_FLOW_ERROR;
}
parse_error:
{
g_free (label);
g_free (proto);
g_set_error (error, GST_WEBRTC_ERROR,
GST_WEBRTC_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 (WebRTCDataChannel * channel, GBytes * data)
{
gst_webrtc_data_channel_on_message_data (GST_WEBRTC_DATA_CHANNEL (channel),
data);
}
static void
_emit_have_string (GstWebRTCDataChannel * channel, gchar * str)
{
gst_webrtc_data_channel_on_message_string (GST_WEBRTC_DATA_CHANNEL (channel),
str);
}
static GstFlowReturn
_data_channel_have_sample (WebRTCDataChannel * 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_ERROR,
GST_WEBRTC_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_ERROR,
GST_WEBRTC_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_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE,
"Failed to map received buffer");
ret = GST_FLOW_ERROR;
} else {
ret = _parse_control_packet (channel, info.data, info.size, error);
gst_buffer_unmap (buffer, &info);
}
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_ERROR,
GST_WEBRTC_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);
gst_buffer_unmap (buffer, &info);
}
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_ERROR,
GST_WEBRTC_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_ERROR,
GST_WEBRTC_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)
{
WebRTCDataChannel *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)
{
WebRTCDataChannel *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
webrtc_data_channel_start_negotiation (WebRTCDataChannel * channel)
{
GstBuffer *buffer;
g_return_if_fail (!channel->parent.negotiated);
g_return_if_fail (channel->parent.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->parent.id,
channel->parent.label, channel->parent.protocol,
channel->parent.ordered ? "true" : "false");
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
channel->parent.buffered_amount += gst_buffer_get_size (buffer);
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (&channel->parent), "buffered-amount");
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_ERROR,
GST_WEBRTC_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 (WebRTCDataChannel * channel,
GstSctpSendMetaPartiallyReliability * reliability, guint * rel_param)
{
if (channel->parent.max_retransmits != -1) {
*reliability = GST_SCTP_SEND_META_PARTIAL_RELIABILITY_RTX;
*rel_param = channel->parent.max_retransmits;
} else if (channel->parent.max_packet_lifetime != -1) {
*reliability = GST_SCTP_SEND_META_PARTIAL_RELIABILITY_TTL;
*rel_param = channel->parent.max_packet_lifetime;
} else {
*reliability = GST_SCTP_SEND_META_PARTIAL_RELIABILITY_NONE;
*rel_param = 0;
}
}
static gboolean
_is_within_max_message_size (WebRTCDataChannel * channel, gsize size)
{
return size <= channel->sctp_transport->max_message_size;
}
static void
webrtc_data_channel_send_data (GstWebRTCDataChannel * base_channel,
GBytes * bytes)
{
WebRTCDataChannel *channel = WEBRTC_DATA_CHANNEL (base_channel);
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_ERROR,
GST_WEBRTC_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->parent.ordered,
reliability, rel_param);
GST_LOG_OBJECT (channel, "Sending data using buffer %" GST_PTR_FORMAT,
buffer);
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
channel->parent.buffered_amount += gst_buffer_get_size (buffer);
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (&channel->parent), "buffered-amount");
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_ERROR,
GST_WEBRTC_ERROR_DATA_CHANNEL_FAILURE, "Failed to send data");
_channel_store_error (channel, error);
_channel_enqueue_task (channel, (ChannelTask) _close_procedure, NULL, NULL);
}
}
static void
webrtc_data_channel_send_string (GstWebRTCDataChannel * base_channel,
const gchar * str)
{
WebRTCDataChannel *channel = WEBRTC_DATA_CHANNEL (base_channel);
GstSctpSendMetaPartiallyReliability reliability;
guint rel_param;
guint32 ppid;
GstBuffer *buffer;
GstFlowReturn ret;
if (!channel->parent.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;
if (!_is_within_max_message_size (channel, size)) {
GError *error = NULL;
g_set_error (&error, GST_WEBRTC_ERROR,
GST_WEBRTC_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;
}
str_copy = g_strdup (str);
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->parent.ordered,
reliability, rel_param);
GST_TRACE_OBJECT (channel, "Sending string using buffer %" GST_PTR_FORMAT,
buffer);
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
channel->parent.buffered_amount += gst_buffer_get_size (buffer);
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (&channel->parent), "buffered-amount");
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_ERROR,
GST_WEBRTC_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,
WebRTCDataChannel * channel)
{
GstWebRTCSCTPTransportState state;
g_object_get (sctp_transport, "state", &state, NULL);
if (state == GST_WEBRTC_SCTP_TRANSPORT_STATE_CONNECTED) {
if (channel->parent.negotiated)
_channel_enqueue_task (channel, (ChannelTask) _emit_on_open, NULL, NULL);
}
}
static void
_on_sctp_notify_state (GObject * sctp_transport, GParamSpec * pspec,
WebRTCDataChannel * channel)
{
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
_on_sctp_notify_state_unlocked (sctp_transport, channel);
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
}
static void
_emit_low_threshold (WebRTCDataChannel * channel, gpointer user_data)
{
gst_webrtc_data_channel_on_buffered_amount_low (GST_WEBRTC_DATA_CHANNEL
(channel));
}
static GstPadProbeReturn
on_appsrc_data (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
{
WebRTCDataChannel *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) {
GST_WEBRTC_DATA_CHANNEL_LOCK (channel);
prev_amount = channel->parent.buffered_amount;
channel->parent.buffered_amount -= size;
GST_TRACE_OBJECT (channel, "checking low-threshold: prev %"
G_GUINT64_FORMAT " low-threshold %" G_GUINT64_FORMAT " buffered %"
G_GUINT64_FORMAT, prev_amount,
channel->parent.buffered_amount_low_threshold,
channel->parent.buffered_amount);
if (prev_amount >= channel->parent.buffered_amount_low_threshold
&& channel->parent.buffered_amount <
channel->parent.buffered_amount_low_threshold) {
_channel_enqueue_task (channel, (ChannelTask) _emit_low_threshold, NULL,
NULL);
}
if (channel->parent.ready_state == GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING
&& channel->parent.buffered_amount <= 0) {
_channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream, NULL,
NULL);
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
g_object_notify (G_OBJECT (&channel->parent), "buffered-amount");
}
return GST_PAD_PROBE_OK;
}
static void
gst_webrtc_data_channel_constructed (GObject * object)
{
WebRTCDataChannel *channel = 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)
{
WebRTCDataChannel *channel = 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;
}
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_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
webrtc_data_channel_class_init (WebRTCDataChannelClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstWebRTCDataChannelClass *channel_class =
(GstWebRTCDataChannelClass *) klass;
gobject_class->constructed = gst_webrtc_data_channel_constructed;
gobject_class->finalize = gst_webrtc_data_channel_finalize;
channel_class->send_data = webrtc_data_channel_send_data;
channel_class->send_string = webrtc_data_channel_send_string;
channel_class->close = webrtc_data_channel_close;
}
static void
webrtc_data_channel_init (WebRTCDataChannel * channel)
{
}
static void
_data_channel_set_sctp_transport (WebRTCDataChannel * channel,
WebRTCSCTPTransport * sctp)
{
g_return_if_fail (GST_IS_WEBRTC_DATA_CHANNEL (channel));
g_return_if_fail (GST_IS_WEBRTC_SCTP_TRANSPORT (sctp));
GST_WEBRTC_DATA_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_stream_reset),
channel);
g_signal_connect (sctp, "notify::state", G_CALLBACK (_on_sctp_notify_state),
channel);
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
}
void
webrtc_data_channel_link_to_sctp (WebRTCDataChannel * channel,
WebRTCSCTPTransport * 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);
_on_sctp_notify_state_unlocked (G_OBJECT (sctp_transport), channel);
}
}
}
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