webrtcbin: rtcp codec stats are messed up

i'm regularily querying the RTCP stats of my webrtcbin using the "get-stats" signal. the reply of the promise is supposed to look kinda like this (transformed into JSON):

{
   "stats": {
     "codec-stats-src_0": {
       "clock-rate": 48000,
       "id": "codec-stats-src_0",
       "payload-type": 111,
       "timestamp": 12445409.612,
       "type": "GST_WEBRTC_STATS_CODEC"
     },
     "codec-stats-src_1": {
       "clock-rate": 90000,
       "id": "codec-stats-src_1",
       "payload-type": 100,
       "timestamp": 12445409.612,
       "type": "GST_WEBRTC_STATS_CODEC"
     },
     "ice-candidate-pair_webrtcnicetransport0": {
       "id": "ice-candidate-pair_webrtcnicetransport0",
       "timestamp": 12445409.612,
       "type": "GST_WEBRTC_STATS_TRANSPORT"
     },
     "ice-candidate-pair_webrtcnicetransport2": {
       "id": "ice-candidate-pair_webrtcnicetransport2",
       "timestamp": 12445409.612,
       "type": "GST_WEBRTC_STATS_TRANSPORT"
     },
     "peer-connection-stats": {
       "data-channels-accepted": 0,
       "data-channels-closed": 0,
       "data-channels-opened": 0,
       "data-channels-requested": 0,
       "id": "peer-connection-stats",
       "timestamp": 12445409.612,
       "type": "GST_WEBRTC_STATS_PEER_CONNECTION"
     },
     "rtp-inbound-stream-stats_2001382111": {
       "bytes-received": 157941,
       "codec-id": "codec-stats-src_1",
       "fir-count": 0,
       "id": "rtp-inbound-stream-stats_2001382111",
       "jitter": "inf",
       "nack-count": 0,
       "packets-lost": 0,
       "packets-received": 202,
       "pli-count": 0,
       "remote-id": "rtp-remote-outbound-stream-stats_2001382111",
       "ssrc": 2001382111,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport2",
       "type": "GST_WEBRTC_STATS_INBOUND_RTP"
     },
     "rtp-inbound-stream-stats_2060799192": {
       "bytes-received": 20752,
       "codec-id": "codec-stats-src_0",
       "fir-count": 0,
       "id": "rtp-inbound-stream-stats_2060799192",
       "jitter": "inf",
       "nack-count": 0,
       "packets-lost": 0,
       "packets-received": 214,
       "pli-count": 0,
       "remote-id": "rtp-remote-outbound-stream-stats_2060799192",
       "ssrc": 2060799192,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport0",
       "type": "GST_WEBRTC_STATS_INBOUND_RTP"
     },
     "rtp-outbound-stream-stats_2001382111": {
       "bytes-sent": 0,
       "codec-id": "codec-stats-src_1",
       "fir-count": 1,
       "id": "rtp-outbound-stream-stats_2001382111",
       "nack-count": 0,
       "packets-sent": 0,
       "pli-count": 0,
       "remote-id": "rtp-remote-inbound-stream-stats_2001382111",
       "ssrc": 2001382111,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport2",
       "type": "GST_WEBRTC_STATS_OUTBOUND_RTP"
     },
     "rtp-outbound-stream-stats_2060799192": {
       "bytes-sent": 0,
       "codec-id": "codec-stats-src_0",
       "fir-count": 0,
       "id": "rtp-outbound-stream-stats_2060799192",
       "nack-count": 0,
       "packets-sent": 0,
       "pli-count": 0,
       "remote-id": "rtp-remote-inbound-stream-stats_2060799192",
       "ssrc": 2060799192,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport0",
       "type": "GST_WEBRTC_STATS_OUTBOUND_RTP"
     },
     "rtp-remote-inbound-stream-stats_2001382111": {
       "codec-id": "codec-stats-src_1",
       "id": "rtp-remote-inbound-stream-stats_2001382111",
       "jitter": 0.0,
       "local-id": "rtp-outbound-stream-stats_2001382111",
       "packets-lost": 0,
       "round-trip-time": 0.0,
       "ssrc": 2001382111,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport2",
       "type": "GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
     },
     "rtp-remote-inbound-stream-stats_2060799192": {
       "codec-id": "codec-stats-src_0",
       "id": "rtp-remote-inbound-stream-stats_2060799192",
       "jitter": 0.0,
       "local-id": "rtp-outbound-stream-stats_2060799192",
       "packets-lost": 0,
       "round-trip-time": 0.0,
       "ssrc": 2060799192,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport0",
       "type": "GST_WEBRTC_STATS_REMOTE_INBOUND_RTP"
     },
     "rtp-remote-outbound-stream-stats_2001382111": {
       "codec-id": "codec-stats-src_1",
       "id": "rtp-remote-outbound-stream-stats_2001382111",
       "local-id": "rtp-inbound-stream-stats_2001382111",
       "ssrc": 2001382111,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport2",
       "type": "GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
     },
     "rtp-remote-outbound-stream-stats_2060799192": {
       "codec-id": "codec-stats-src_0",
       "id": "rtp-remote-outbound-stream-stats_2060799192",
       "local-id": "rtp-inbound-stream-stats_2060799192",
       "ssrc": 2060799192,
       "timestamp": 12445409.612,
       "transport-id": "transport-stats_webrtcdtlstransport0",
       "type": "GST_WEBRTC_STATS_REMOTE_OUTBOUND_RTP"
     },
     "transport-stats_webrtcdtlstransport0": {
       "id": "transport-stats_webrtcdtlstransport0",
       "timestamp": 12445409.612,
       "type": "GST_WEBRTC_STATS_TRANSPORT"
     },
     "transport-stats_webrtcdtlstransport2": {
       "id": "transport-stats_webrtcdtlstransport2",
       "timestamp": 12445409.612,
       "type": "GST_WEBRTC_STATS_TRANSPORT"
     }
   },
   "type": "stats"
 }

starting with commit 9f684a2f (webrtcbin: implement support for group: BUNDLE) this changes in a way where all stream stats for every stream are still listed and counted seperately but all have either have the "codec-id": "codec-stats-src_1" or _0. therefore it's hard to distinguish between audio and video packets and it looks like a bug to me.