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webrtcbin: Set transceiver kind and codec preferences immediately when creating it

Browse source 
Otherwise the on-new-transceiver signal will always be emitted with kind
set to UNKNOWN and no codec preferences although both are often known at
this point already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2360>
This commit is contained in:
Sebastian Dröge 2021-06-25 10:20:06 +03:00
parent dcc49f846b
commit 096a7f1ac0
2 changed files with 28 additions and 22 deletions
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46
ext/webrtc/gstwebrtcbin.c
View file

@ -2025,7 +2025,8 @@ gst_webrtc_bin_attach_tos (GstWebRTCBin * webrtc)
static WebRTCTransceiver *
_create_webrtc_transceiver (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiverDirection direction, guint mline)
GstWebRTCRTPTransceiverDirection direction, guint mline, GstWebRTCKind kind,
GstCaps * codec_preferences)
{
WebRTCTransceiver *trans;
GstWebRTCRTPTransceiver *rtp_trans;
@ -2038,6 +2039,9 @@ _create_webrtc_transceiver (GstWebRTCBin * webrtc,
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
rtp_trans->direction = direction;
rtp_trans->mline = mline;
rtp_trans->kind = kind;
rtp_trans->codec_preferences =
codec_preferences ? gst_caps_ref (codec_preferences) : NULL;
/* FIXME: We don't support stopping transceiver yet so they're always not stopped */
rtp_trans->stopped = FALSE;
@ -3712,8 +3716,14 @@ _create_answer_task (GstWebRTCBin * webrtc, const GstStructure * options,
if (!rtp_trans) {
GstCaps *trans_caps;
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
trans = _create_webrtc_transceiver (webrtc, answer_dir, i);
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0)
kind = GST_WEBRTC_KIND_AUDIO;
else
kind = GST_WEBRTC_KIND_VIDEO;
trans = _create_webrtc_transceiver (webrtc, answer_dir, i, kind, NULL);
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT
@ -4859,10 +4869,10 @@ _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) {
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
/* No existing transceiver, find an unused one */
if (!trans) {
GstWebRTCKind kind;
if (g_strcmp0 (gst_sdp_media_get_media (media), "audio") == 0)
kind = GST_WEBRTC_KIND_AUDIO;
else
@ -4879,7 +4889,7 @@ _update_transceivers_from_sdp (GstWebRTCBin * webrtc, SDPSource source,
* a default value when the transceiver is created internally */
if (!trans) {
WebRTCTransceiver *t = _create_webrtc_transceiver (webrtc,
_get_direction_from_media (media), i);
_get_direction_from_media (media), i, kind, NULL);
webrtc_transceiver_set_transport (t, stream);
trans = GST_WEBRTC_RTP_TRANSCEIVER (t);
}
@ -5678,25 +5688,18 @@ gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
GstWebRTCRTPTransceiverDirection direction, GstCaps * caps)
{
WebRTCTransceiver *trans;
GstWebRTCRTPTransceiver *rtp_trans;
g_return_val_if_fail (direction != GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_NONE,
NULL);
PC_LOCK (webrtc);
trans = _create_webrtc_transceiver (webrtc, direction, -1);
trans =
_create_webrtc_transceiver (webrtc, direction, -1,
webrtc_kind_from_caps (caps), caps);
GST_LOG_OBJECT (webrtc,
"Created new unassociated transceiver %" GST_PTR_FORMAT, trans);
rtp_trans = GST_WEBRTC_RTP_TRANSCEIVER (trans);
if (caps) {
GST_OBJECT_LOCK (trans);
rtp_trans->codec_preferences = gst_caps_ref (caps);
GST_OBJECT_UNLOCK (trans);
_update_transceiver_kind_from_caps (rtp_trans, caps);
}
PC_UNLOCK (webrtc);
return gst_object_ref (trans);
@ -6631,7 +6634,6 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
GstWebRTCKind kind = GST_WEBRTC_KIND_UNKNOWN;
guint i;
if (caps)
kind = webrtc_kind_from_caps (caps);
for (i = 0; i < webrtc->priv->transceivers->len; i++) {
@ -6677,17 +6679,21 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
if (!trans) {
trans = GST_WEBRTC_RTP_TRANSCEIVER (_create_webrtc_transceiver (webrtc,
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, -1));
GST_WEBRTC_RTP_TRANSCEIVER_DIRECTION_SENDRECV, -1,
webrtc_kind_from_caps (caps), NULL));
GST_LOG_OBJECT (webrtc, "Created new transceiver %" GST_PTR_FORMAT, trans);
} else {
GST_LOG_OBJECT (webrtc, "Using existing transceiver %" GST_PTR_FORMAT
" for mline %u", trans, serial);
if (caps) {
if (!_update_transceiver_kind_from_caps (trans, caps))
GST_WARNING_OBJECT (webrtc,
"Trying to change transceiver %d kind from %d to %d",
serial, trans->kind, webrtc_kind_from_caps (caps));
}
}
pad = _create_pad_for_sdp_media (webrtc, GST_PAD_SINK, trans, serial);
if (caps)
_update_transceiver_kind_from_caps (trans, caps);
pad->block_id = gst_pad_add_probe (GST_PAD (pad), GST_PAD_PROBE_TYPE_BLOCK |
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
(GstPadProbeCallback) sink_pad_block, NULL, NULL);

2
ext/webrtc/utils.c
View file

@ -212,7 +212,7 @@ webrtc_kind_from_caps (const GstCaps * caps)
GstStructure *s;
const gchar *media;
if (gst_caps_get_size (caps) == 0)
if (!caps || gst_caps_get_size (caps) == 0)
return GST_WEBRTC_KIND_UNKNOWN;
s = gst_caps_get_structure (caps, 0);