diff options
| -rw-r--r-- | subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c | 27 |
1 files changed, 18 insertions, 9 deletions
diff --git a/subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c b/subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c index 85e8d23970..1305c740ac 100644 --- a/subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c +++ b/subprojects/gst-plugins-bad/ext/webrtc/webrtcdatachannel.c @@ -437,11 +437,15 @@ _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); @@ -1045,6 +1049,16 @@ 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) { @@ -1063,11 +1077,6 @@ 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"); } |