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webrtc: Unblock transportreceivebin for send-only bundled streams

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If there is any active mline in a bundle, we need to unblock
the transportreceivebin for DTLS setup and RTCP reception,
otherwise no data can ever start flowing.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/1206
This commit is contained in:
Jan Schmidt 2020-03-03 03:39:50 +11:00 committed by Sebastian Dröge
parent cb48733ff3
commit 905988c63f
2 changed files with 51 additions and 18 deletions
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68
ext/webrtc/gstwebrtcbin.c
View file

@ -3718,7 +3718,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
const GstSDPMedia *media = gst_sdp_message_get_media (sdp, media_idx);
GstWebRTCDTLSSetup new_setup;
gboolean new_rtcp_mux, new_rtcp_rsize;
ReceiveState receive_state = 0;
ReceiveState receive_state = RECEIVE_STATE_UNSET;
int i;
rtp_trans->mline = media_idx;
@ -3804,7 +3804,20 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
gst_object_unref (pad);
}
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;
}
/* XXX: send eos event up the sink pad as well? */
} 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. */
receive_state = RECEIVE_STATE_PASS;
stream->active = TRUE;
}
if (new_dir == GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDONLY ||
@ -3860,7 +3873,6 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
}
}
receive_state = RECEIVE_STATE_PASS;
rtp_trans->mline = media_idx;
rtp_trans->current_direction = new_dir;
@ -3878,7 +3890,7 @@ _update_transceiver_from_sdp_media (GstWebRTCBin * webrtc,
/* 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 > 0)
if (receive_state != RECEIVE_STATE_UNSET)
transport_receive_bin_set_receive_state (stream->receive_bin,
receive_state);
}
@ -4079,6 +4091,8 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
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 (!_parse_bundle (sdp->sdp, &bundled))
goto done;
@ -4092,9 +4106,15 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
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);
}
@ -4121,6 +4141,8 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
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);
}
@ -4134,24 +4156,25 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
} else {
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0 ||
g_strcmp0 (gst_sdp_media_get_media (media), "video") == 0) {
if (trans) {
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
trans, bundled, bundle_idx);
} else {
/* No existing transceiver, find an unused one */
if (!trans) {
trans = _find_transceiver (webrtc, NULL,
(FindTransceiverFunc) _find_compatible_unassociated_transceiver);
/* XXX: default to the advertised direction in the sdp for new
* transceviers. 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) {
trans =
GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
_get_direction_from_media (media), i));
}
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
trans, bundled, bundle_idx);
}
/* 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) {
trans =
GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
_get_direction_from_media (media), i));
}
_update_transceiver_from_sdp_media (webrtc, sdp->sdp, i, stream,
trans, bundled, bundle_idx);
} else if (_message_media_is_datachannel (sdp->sdp, i)) {
_update_data_channel_from_sdp_media (webrtc, sdp->sdp, i, stream);
} else {
@ -4160,6 +4183,15 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
}
}
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:

1
ext/webrtc/transportstream.h
View file

@ -52,6 +52,7 @@ struct _TransportStream
gboolean rtcp_mux;
gboolean rtcp_rsize;
gboolean dtls_client;
gboolean active; /* TRUE if any mline in the bundle/transport is active */
TransportSendBin *send_bin; /* bin containing all the sending transport elements */
TransportReceiveBin *receive_bin; /* bin containing all the receiving transport elements */
GstWebRTCICEStream *stream;