rtspsrc just stopping playback

[saepia]

Hello,

I encounter odd behaviour with rtspsrc.

I have Gst-backed RTSP server that encodes opus audio stream and rtspsrc on the other side (on Android) that tries to play it back.

Generally speaking it works fine but it just stops playback after some time (5-10 min) with no error, warning, message etc. (I am logging everything that appears on the pipeline's bus).

When I restart the pipeline it just starts to work again so I assume that problem is on the receiver side.

The receiver's pipeline is

rtspsrc location=rtsp://.... user-agent=... drop-on-latency=true latency=500 tls-database=... protocols=... username=... password=... ! decodebin ! audioconvert ! audioresample ! queue2 ! openslessink

(protocols are  0x00000001 | 0x00000002 | 0x00000004 | 0x00000010 | 0x00000020)

I am using 1.8.0 on both sides.

I am not sure if this is related but during playback I ocassionally get

04-11 16:38:41.989 21543 21629 W GStreamer+audiobasesink: 0:04:35.920673280 0xb8e045b0 gstaudiobasesink.c:1484:gst_audio_base_sink_skew_slaving:<audio_interface_playback> correct clock skew +0:00:00.020063566 > +0:00:00.020000000
04-11 16:38:42.006 21543 21629 W GStreamer+audiobasesink: 0:04:35.938068176 0xb8e045b0 gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<audio_interface_playback> correct clock skew -0:00:00.020290466 < -+0:00:00.020000000

Any suggestions what can be the reason for such mysteroius hangs?

How can I enable equivalent of GST_DEBUG env var on Android?

m.

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[saepia]

I was able to reproduce the same behaviour on mac os x. I've used gst-launch -m -vv to run identical pipeline and the only output was

/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstRtpBin:manager/GstRtpSession:rtpsession0: stats = "application/x-rtp-session-stats\,\ rtx-drop-count\=\(uint\)0\,\ sent-nack-count\=\(uint\)0\,\ recv-nack-count\=\(uint\)0\,\ source-stats\=\(GValueArray\)NULL\,\ rtx-count\=\(uint\)0\;"

(repeated many times)

and it kept appearing even when audio was not playing

am I doing something wrong or is it a bug?

m.

2016-04-11 16:48 GMT+02:00 [hidden email] <[hidden email]>:

Hello,

I encounter odd behaviour with rtspsrc.

I have Gst-backed RTSP server that encodes opus audio stream and rtspsrc on the other side (on Android) that tries to play it back.

Generally speaking it works fine but it just stops playback after some time (5-10 min) with no error, warning, message etc. (I am logging everything that appears on the pipeline's bus).

When I restart the pipeline it just starts to work again so I assume that problem is on the receiver side.

The receiver's pipeline is

rtspsrc location=rtsp://.... user-agent=... drop-on-latency=true latency=500 tls-database=... protocols=... username=... password=... ! decodebin ! audioconvert ! audioresample ! queue2 ! openslessink

(protocols are  0x00000001 | 0x00000002 | 0x00000004 | 0x00000010 | 0x00000020)

I am using 1.8.0 on both sides.

I am not sure if this is related but during playback I ocassionally get

04-11 16:38:41.989 21543 21629 W GStreamer+audiobasesink: 0:04:35.920673280 0xb8e045b0 gstaudiobasesink.c:1484:gst_audio_base_sink_skew_slaving:<audio_interface_playback> correct clock skew +0:00:00.020063566 > +0:00:00.020000000
04-11 16:38:42.006 21543 21629 W GStreamer+audiobasesink: 0:04:35.938068176 0xb8e045b0 gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<audio_interface_playback> correct clock skew -0:00:00.020290466 < -+0:00:00.020000000

Any suggestions what can be the reason for such mysteroius hangs?

How can I enable equivalent of GST_DEBUG env var on Android?

m.

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[saepia]

The same happens with playbin. I have run it with GST_DEBUG=*:4,rtspsrc:5,opus:5 and there was no output prior to hang.

m.

2016-04-11 17:27 GMT+02:00 [hidden email] <[hidden email]>:

I was able to reproduce the same behaviour on mac os x. I've used gst-launch -m -vv to run identical pipeline and the only output was

/GstPipeline:pipeline0/GstRTSPSrc:rtspsrc0/GstRtpBin:manager/GstRtpSession:rtpsession0: stats = "application/x-rtp-session-stats\,\ rtx-drop-count\=\(uint\)0\,\ sent-nack-count\=\(uint\)0\,\ recv-nack-count\=\(uint\)0\,\ source-stats\=\(GValueArray\)NULL\,\ rtx-count\=\(uint\)0\;"

(repeated many times)

and it kept appearing even when audio was not playing

am I doing something wrong or is it a bug?

m.

2016-04-11 16:48 GMT+02:00 [hidden email] <[hidden email]>:

Hello,

I encounter odd behaviour with rtspsrc.

I have Gst-backed RTSP server that encodes opus audio stream and rtspsrc on the other side (on Android) that tries to play it back.

Generally speaking it works fine but it just stops playback after some time (5-10 min) with no error, warning, message etc. (I am logging everything that appears on the pipeline's bus).

When I restart the pipeline it just starts to work again so I assume that problem is on the receiver side.

The receiver's pipeline is

rtspsrc location=rtsp://.... user-agent=... drop-on-latency=true latency=500 tls-database=... protocols=... username=... password=... ! decodebin ! audioconvert ! audioresample ! queue2 ! openslessink

(protocols are  0x00000001 | 0x00000002 | 0x00000004 | 0x00000010 | 0x00000020)

I am using 1.8.0 on both sides.

I am not sure if this is related but during playback I ocassionally get

04-11 16:38:41.989 21543 21629 W GStreamer+audiobasesink: 0:04:35.920673280 0xb8e045b0 gstaudiobasesink.c:1484:gst_audio_base_sink_skew_slaving:<audio_interface_playback> correct clock skew +0:00:00.020063566 > +0:00:00.020000000
04-11 16:38:42.006 21543 21629 W GStreamer+audiobasesink: 0:04:35.938068176 0xb8e045b0 gstaudiobasesink.c:1512:gst_audio_base_sink_skew_slaving:<audio_interface_playback> correct clock skew -0:00:00.020290466 < -+0:00:00.020000000

Any suggestions what can be the reason for such mysteroius hangs?

How can I enable equivalent of GST_DEBUG env var on Android?

m.

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[Sebastian Dröge-3]

On Mo, 2016-04-11 at 16:48 +0200, [hidden email] wrote:

> Hello,
>
> I encounter odd behaviour with rtspsrc.
>
> I have Gst-backed RTSP server that encodes opus audio stream and
> rtspsrc on the other side (on Android) that tries to play it back.
>
> Generally speaking it works fine but it just stops playback after
> some time (5-10 min) with no error, warning, message etc. (I am
> logging everything that appears on the pipeline's bus).
>
> When I restart the pipeline it just starts to work again so I assume
> that problem is on the receiver side.
>
> The receiver's pipeline is 
>
> rtspsrc location=rtsp://.... user-agent=... drop-on-latency=true
> latency=500 tls-database=... protocols=... username=... password=...
> ! decodebin ! audioconvert ! audioresample ! queue2 ! openslessink
>
> (protocols are  0x00000001 | 0x00000002 | 0x00000004 | 0x00000010 |
> 0x00000020)
>
> I am using 1.8.0 on both sides.

That seems like a bug to me, it should work fine.

Using queue2 here is not useful though, a normal queue is better. But
that should be unrelated.

> uggestions what can be the reason for such mysteroius hangs?
>
> How can I enable equivalent of GST_DEBUG env var on Android?

The environment variable is just convenience around the gst_debug_set*
API, e.g. gst_debug_set_threshold_from_string("*:6", TRUE).

--
Sebastian Dröge, Centricular Ltd · http://www.centricular.com


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[Sebastian Dröge-3]

On Di, 2016-04-12 at 09:53 +0300, Sebastian Dröge wrote:
> 
> > I am using 1.8.0 on both sides.
> That seems like a bug to me, it should work fine.
>
> Using queue2 here is not useful though, a normal queue is better. But
> that should be unrelated.

It was reported already:
  https://bugzilla.gnome.org/show_bug.cgi?id=764905

--
Sebastian Dröge, Centricular Ltd · http://www.centricular.com


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[saepia]

Using queue2 here is not useful though, a normal queue is better. But
that should be unrelated.

In what sense queue is better than queue2?

m.

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[Sebastian Dröge-3]

On Di, 2016-04-12 at 12:52 +0200, [hidden email] wrote:
>
> > Using queue2 here is not useful though, a normal queue is better.
> > But
> > that should be unrelated.
> >
>
> In what sense queue is better than queue2?

They're not doing the same thing, queue2 is not queue version 2. queue2
is for buffering of network streams mostly.

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Sebastian Dröge, Centricular Ltd · http://www.centricular.com


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