webrtcbin get-stats don't contain any rtpstats.

[Faraz Khan]

For posterity the problem was that I was using gstreamer 1.16 where this
stuff isn't linked up. All works fine using master. Thanks for the
webrtcbin folks!

On Wed, Oct 7, 2020 at 7:30 AM Faraz Khan <farazrkhan at gmail.com> wrote:

> Gotcha. This issue made me think otherwise. Seems like these guys
> (gstreamer devs) are getting way more stats than me:
>
> https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/889
>
>
> On Tue, Oct 6, 2020, 11:29 PM Matthew Waters <ystreet00 at gmail.com> wrote:
>
>> No, you don't 'need' to do anything for rtpbin to expose the stats, they
>> just aren't always exposed in the webrtcbin stats as the webrtcbin stats
>> are incomplete.
>>
>> Cheers
>> -Matt
>>
>> On 7/10/20 5:06 pm, Faraz Khan wrote:
>>
>> 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
>>>
>>>
>>>
>>
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