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webrtc: do not tear down data channel before data is flushed

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Current implementation can in some cases detect
that all data is sent but in reality it is not,
leading to a push to an unlinked pad.
This is a race between the probe used to track data sent and a
call to close.

This patch sends an EOS before starting the close procedure
and then waits for the EOS event to come through to the
src pad before commencing with tear down.
This ensures that any queued data before EOS is flushed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4462>
This commit is contained in:
Daniel Moberg 2023-04-18 11:21:05 +02:00 committed by GStreamer Marge Bot
parent 262be30b6f
commit 4c1de25e9d
1 changed files with 18 additions and 9 deletions
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27
subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c
View file

@ -427,11 +427,15 @@ _close_procedure (WebRTCDataChannel * channel, gpointer user_data)
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);
/* Make sure that all data enqueued gets properly sent before data channel is closed. */
GstFlowReturn ret =
gst_app_src_end_of_stream (GST_APP_SRC (WEBRTC_DATA_CHANNEL
(channel)->appsrc));
if (ret != GST_FLOW_OK) {
GST_WARNING_OBJECT (channel, "Send end of stream returned %i, %s", ret,
gst_flow_get_name (ret));
}
return;
}
GST_WEBRTC_DATA_CHANNEL_UNLOCK (channel);
@ -1035,6 +1039,16 @@ on_appsrc_data (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
} 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);
} else if (GST_PAD_PROBE_INFO_TYPE (info) &
GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM) {
GstEvent *event = GST_PAD_PROBE_INFO_EVENT (info);
if (GST_EVENT_TYPE (event) == GST_EVENT_EOS
&& channel->parent.ready_state ==
GST_WEBRTC_DATA_CHANNEL_STATE_CLOSING) {
_channel_enqueue_task (channel, (ChannelTask) _close_sctp_stream, NULL,
NULL);
return GST_PAD_PROBE_DROP;
}
}
if (size > 0) {
@ -1053,11 +1067,6 @@ on_appsrc_data (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
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");
}