webrtcbin: Hold transceiver lock when accessing codec_preferences

This is required to allow the applications to modify the preferences.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2183>
This commit is contained in:
Olivier Crête 2021-04-30 15:04:33 -04:00
parent e9f14ed117
commit dc6655542d

View file

@ -1505,66 +1505,69 @@ _find_codec_preferences (GstWebRTCBin * webrtc,
{
WebRTCTransceiver *trans = (WebRTCTransceiver *) rtp_trans;
GstCaps *ret = NULL;
GstWebRTCBinPad *pad = NULL;
GST_LOG_OBJECT (webrtc, "retrieving codec preferences from %" GST_PTR_FORMAT,
trans);
if (rtp_trans && rtp_trans->codec_preferences) {
GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT,
rtp_trans->codec_preferences);
ret = gst_caps_ref (rtp_trans->codec_preferences);
} else {
GstWebRTCBinPad *pad = NULL;
/* try to find a pad */
if (!trans
|| !(pad = _find_pad_for_transceiver (webrtc, direction, rtp_trans)))
pad = _find_pad_for_mline (webrtc, direction, media_idx);
if (!pad) {
if (trans && trans->last_configured_caps)
ret = gst_caps_ref (trans->last_configured_caps);
} else {
GstCaps *caps = NULL;
if (pad->received_caps) {
caps = gst_caps_ref (pad->received_caps);
} else if ((caps = gst_pad_get_current_caps (GST_PAD (pad)))) {
GST_LOG_OBJECT (webrtc, "Using current pad caps: %" GST_PTR_FORMAT,
caps);
} else {
static GstStaticCaps static_filter =
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) { audio, video }, payload = (int) [ 0, 127 ]");
GstCaps *filter = gst_static_caps_get (&static_filter);
filter = gst_caps_make_writable (filter);
if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "audio", NULL);
else if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "video", NULL);
caps = gst_pad_peer_query_caps (GST_PAD (pad), filter);
GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT,
caps);
if (!gst_caps_is_fixed (caps) || gst_caps_is_equal (caps, filter)
|| gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
gst_caps_unref (caps);
caps = NULL;
}
gst_caps_unref (filter);
}
if (caps) {
if (trans)
gst_caps_replace (&trans->last_configured_caps, caps);
ret = caps;
}
gst_object_unref (pad);
if (rtp_trans) {
GST_OBJECT_LOCK (rtp_trans);
if (rtp_trans->codec_preferences) {
GST_LOG_OBJECT (webrtc, "Using codec preferences: %" GST_PTR_FORMAT,
rtp_trans->codec_preferences);
ret = gst_caps_ref (rtp_trans->codec_preferences);
}
GST_OBJECT_UNLOCK (rtp_trans);
if (ret)
return ret;
}
/* try to find a pad */
if (!trans
|| !(pad = _find_pad_for_transceiver (webrtc, direction, rtp_trans)))
pad = _find_pad_for_mline (webrtc, direction, media_idx);
if (!pad) {
if (trans && trans->last_configured_caps)
ret = gst_caps_ref (trans->last_configured_caps);
} else {
GstCaps *caps = NULL;
if (pad->received_caps) {
caps = gst_caps_ref (pad->received_caps);
} else if ((caps = gst_pad_get_current_caps (GST_PAD (pad)))) {
GST_LOG_OBJECT (webrtc, "Using current pad caps: %" GST_PTR_FORMAT, caps);
} else {
static GstStaticCaps static_filter =
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) { audio, video }, payload = (int) [ 0, 127 ]");
GstCaps *filter = gst_static_caps_get (&static_filter);
filter = gst_caps_make_writable (filter);
if (rtp_trans->kind == GST_WEBRTC_KIND_AUDIO)
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "audio", NULL);
else if (rtp_trans->kind == GST_WEBRTC_KIND_VIDEO)
gst_caps_set_simple (filter, "media", G_TYPE_STRING, "video", NULL);
caps = gst_pad_peer_query_caps (GST_PAD (pad), filter);
GST_LOG_OBJECT (webrtc, "Using peer query caps: %" GST_PTR_FORMAT, caps);
if (!gst_caps_is_fixed (caps) || gst_caps_is_equal (caps, filter)
|| gst_caps_is_empty (caps) || gst_caps_is_any (caps)) {
gst_caps_unref (caps);
caps = NULL;
}
gst_caps_unref (filter);
}
if (caps) {
if (trans)
gst_caps_replace (&trans->last_configured_caps, caps);
ret = caps;
}
gst_object_unref (pad);
}
if (!ret)
@ -2697,6 +2700,7 @@ gather_reserved_pts (GstWebRTCBin * webrtc)
GstWebRTCRTPTransceiver *trans;
trans = g_ptr_array_index (webrtc->priv->transceivers, i);
GST_OBJECT_LOCK (trans);
if (trans->codec_preferences) {
guint j, n;
gint pt;
@ -2711,6 +2715,7 @@ gather_reserved_pts (GstWebRTCBin * webrtc)
}
}
}
GST_OBJECT_UNLOCK (trans);
}
GST_OBJECT_UNLOCK (webrtc);
@ -5497,7 +5502,9 @@ gst_webrtc_bin_add_transceiver (GstWebRTCBin * webrtc,
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);
}
@ -6403,14 +6410,17 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
}
if (caps) {
GST_OBJECT_LOCK (trans);
if (trans->codec_preferences &&
!gst_caps_can_intersect (caps, trans->codec_preferences)) {
GST_ERROR_OBJECT (element, "Tried to request a new sink pad %s for"
" existing m-line %d, but requested caps %" GST_PTR_FORMAT
" don't match existing codec preferences %" GST_PTR_FORMAT,
name, serial, caps, trans->codec_preferences);
GST_OBJECT_UNLOCK (trans);
goto error_out;
}
GST_OBJECT_UNLOCK (trans);
if (trans->kind != GST_WEBRTC_KIND_UNKNOWN) {
GstWebRTCKind kind = webrtc_kind_from_caps (caps);
@ -6439,6 +6449,7 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
GstWebRTCRTPTransceiver *tmptrans =
g_ptr_array_index (webrtc->priv->transceivers, i);
GstWebRTCBinPad *pad2;
gboolean has_matching_caps;
/* Ignore transceivers with a non-matching kind */
if (tmptrans->kind != GST_WEBRTC_KIND_UNKNOWN &&
@ -6462,11 +6473,13 @@ gst_webrtc_bin_request_new_pad (GstElement * element, GstPadTemplate * templ,
continue;
}
GST_OBJECT_LOCK (tmptrans);
has_matching_caps = (caps && tmptrans->codec_preferences &&
!gst_caps_can_intersect (caps, tmptrans->codec_preferences));
GST_OBJECT_UNLOCK (tmptrans);
/* Ignore transceivers with non-matching caps */
if (caps && tmptrans->codec_preferences &&
!gst_caps_can_intersect (caps, tmptrans->codec_preferences)) {
if (!has_matching_caps)
continue;
}
trans = tmptrans;
break;