webrtcbin get-stats don't contain any rtpstats.

[Faraz Khan]

Oh interesting. Seemed like rtpinputstats (from the RR reports) were
already baked in. I'm specifically looking for fraction lost and
interarrival jitter. Do I have to install a signal for rtcp packets and
extracts those reports myself?

For some reason I thought rtpbin would do this for me.

Thanks!

On Tue, Oct 6, 2020, 10:41 PM Matthew Waters <ystreet00 at gmail.com> wrote:

> Various information provided by the get-stats signal are incomplete.  You
> may have to implement some of the information you need inside webrtcbin
> itself.
>
> Cheers
> -Matt
>
> On 7/10/20 3:25 pm, Faraz Khan wrote:
>
> Hi everyone,
> Gstreamer noob here - but have worked on C/C++ audio/video apps for many
> years now. I have successfully established a webrtc connection with
> electron as the receiver (and gstreamer as send-only) and am now looking to
> implement some sort of simplistic bitrate control and also just to
> understand how gstreamer works.
>
> My first attempt was to get webrtc peerconnection stats. I did:
>
> GstPromise *p = gst_promise_new_with_change_func(_on_stats, NULL, NULL);
> g_signal_emit_by_name(peer->webrtcbin, "get-stats", NULL, p);
>
> And decode them using:
>
> gboolean on_stats_iter(GQuark field, const GValue * value, gpointer pfx) {
>     gchar *str = gst_value_serialize (value);
>     g_print ("%15s: %s", g_quark_to_string (field), str);
>     std::cout << std::endl;
>     g_free (str);
>     return TRUE;
> }
>
> static void _on_stats (GstPromise * promise, gpointer user_data) {
>     const GstStructure *reply = gst_promise_get_reply(promise);
>     gst_structure_foreach(reply, on_stats_iter, (gpointer)reply);
> }
>
> And this generates:
> peer-connection-stats: "peer-connection\,\
> data-channels-opened\=\(uint\)0\,\ data-channels-closed\=\(uint\)0\,\
> data-channels-requested\=\(uint\)0\,\ data-channels-accepted\=\(uint\)0\,\
> type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_PEER_CONNECTION\,\
> timestamp\=\(double\)180230225.79899999\,\
> id\=\(string\)peer-connection-stats\;"
> codec-stats-sink_0: "codec\,\
> type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_CODEC\,\
> timestamp\=\(double\)180230225.79899999\,\
> id\=\(string\)codec-stats-sink_0\,\ payload-type\=\(uint\)96\,\
> clock-rate\=\(uint\)90000\,\ ssrc\=\(uint\)2002222005\;"
> transport-stats_webrtcdtlstransport0: "transport\,\
> type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_TRANSPORT\,\
> timestamp\=\(double\)180230225.79899999\,\
> id\=\(string\)transport-stats_webrtcdtlstransport0\;"
> ice-candidate-pair_webrtcnicetransport0: "transport\,\
> type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_TRANSPORT\,\
> timestamp\=\(double\)180230225.79899999\,\
> id\=\(string\)ice-candidate-pair_webrtcnicetransport0\;"
> codec-stats-sink_1: "codec\,\
> type\=\(GstWebRTCStatsType\)GST_WEBRTC_STATS_CODEC\,\
> timestamp\=\(double\)180230225.79899999\,\
> id\=\(string\)codec-stats-sink_1\,\ payload-type\=\(uint\)97\,\
> clock-rate\=\(uint\)48000\,\ ssrc\=\(uint\)1224562656\;"
>
>
> This has the correct SSRC and codec info - but missing RTP information and
> the data channel stats are incorrect (I do have a open and working data
> channel).
>
> How does one proceed to debug this? What am I missing?
>
>
> _______________________________________________
> gstreamer-devel mailing listgstreamer-devel at lists.freedesktop.orghttps://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://lists.freedesktop.org/archives/gstreamer-devel/attachments/20201006/c20a753a/attachment.htm>