From: Olivier Crête Date: Fri, 30 Apr 2021 19:04:33 +0000 (-0400) Subject: webrtcbin: Hold transceiver lock when accessing codec_preferences X-Git-Tag: 1.19.3~507^2~423 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=dc6655542d49eac55232d69f7e816f7083ae5be7;p=platform%2Fupstream%2Fgstreamer.git webrtcbin: Hold transceiver lock when accessing codec_preferences This is required to allow the applications to modify the preferences. Part-of: --- diff --git a/ext/webrtc/gstwebrtcbin.c b/ext/webrtc/gstwebrtcbin.c index 9ba1850b88..44c1f03da0 100644 --- a/ext/webrtc/gstwebrtcbin.c +++ b/ext/webrtc/gstwebrtcbin.c @@ -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; + 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); - /* 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 (ret) + return ret; + } - if (!pad) { - if (trans && trans->last_configured_caps) - ret = gst_caps_ref (trans->last_configured_caps); + /* 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 { - GstCaps *caps = NULL; + 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); - 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); + 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); - ret = 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); - gst_object_unref (pad); + 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;