Pause/Resume with adder

[avinashgst]

Hi,

I have the pipeline in which need to mix 2 audio at run time.

when single file src1:
filesrc ! decodebin ! audioresample ! audioconvert ! (sink_0) adder !
audioconvert ! autoaudiosink

During runtime adding one more file src2:
filesrc ! decodebin ! audioresample ! audioconvert ! --->
                                                                             
(sink_0) (sink_01) adder ! audioconvert ! autoaudiosink
filesrc ! decodebin ! audioresample ! audioconvert !---->

This is working fine.

To pause src1:
      unlinking audioconvert src and  sink_0 by gst_pad_unlink and called
gst_element_release_request_pad  for sink_0.
      and setting filesrc, decodebin, audioresample, audioconvert to
GST_STATE_PAUSE
Src2 is continue to play and its working as expected.

Now after some time wants to resume the src1 while src2 still playing.
filesrc ! decodebin ! audioresample ! audioconvert ! --->
                                                                             
(sink_01) (sink_02) adder ! audioconvert ! autoaudiosink
filesrc ! decodebin ! audioresample ! audioconvert !---->

          linked audioconvert src and  sink_2 by gst_pad_link and setting
filesrc, decodebin, audioresample, audioconvert to GST_STATE_PLAYING

In that case Src2 also stopped playing and getting underflow error.
From gstreamer log its seems that src1 is not pushing any data after setting
elements to GST_STATE_PLAYING

Any suggestion how to achieve pause/resume operation with adder element.  

Regards,
Avinash


     







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[Stefan Sauer]

On 10/25/2017 01:43 PM, avinashgst wrote:

> Hi,
>
> I have the pipeline in which need to mix 2 audio at run time.
>
> when single file src1:
> filesrc ! decodebin ! audioresample ! audioconvert ! (sink_0) adder !
> audioconvert ! autoaudiosink
>
> During runtime adding one more file src2:
> filesrc ! decodebin ! audioresample ! audioconvert ! --->
>                                                                              
> (sink_0) (sink_01) adder ! audioconvert ! autoaudiosink
> filesrc ! decodebin ! audioresample ! audioconvert !---->
>
> This is working fine.
>
> To pause src1:
>       unlinking audioconvert src and  sink_0 by gst_pad_unlink and called
> gst_element_release_request_pad  for sink_0.
>       and setting filesrc, decodebin, audioresample, audioconvert to
> GST_STATE_PAUSE
> Src2 is continue to play and its working as expected.
>
> Now after some time wants to resume the src1 while src2 still playing.
> filesrc ! decodebin ! audioresample ! audioconvert ! --->
>                                                                              
> (sink_01) (sink_02) adder ! audioconvert ! autoaudiosink
> filesrc ! decodebin ! audioresample ! audioconvert !---->
>
>           linked audioconvert src and  sink_2 by gst_pad_link and setting
> filesrc, decodebin, audioresample, audioconvert to GST_STATE_PLAYING
>
> In that case Src2 also stopped playing and getting underflow error.
> From gstreamer log its seems that src1 is not pushing any data after setting
> elements to GST_STATE_PLAYING
>
> Any suggestion how to achieve pause/resume operation with adder element.  
>
> Regards,
> Avinash
>

If you pause/unpause branches you might want to try liveadder. Also you
don't describe how you re-link src1. Are you using pad-probes? We have
some examples under gst-plugins-base/tests/examples/dynamic/

Stefan
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[filnet]

Le Jeudi 26 octobre 2017 20h35, Stefan Sauer <[hidden email]> a écrit :

On 10/25/2017 01:43 PM, avinashgst wrote:

> Hi,
>
> I have the pipeline in which need to mix 2 audio at run time.
>
> when single file src1:
> filesrc ! decodebin ! audioresample ! audioconvert ! (sink_0) adder !
> audioconvert ! autoaudiosink
>
> During runtime adding one more file src2:
> filesrc ! decodebin ! audioresample ! audioconvert ! --->
>                                                                             
> (sink_0) (sink_01) adder ! audioconvert ! autoaudiosink
> filesrc ! decodebin ! audioresample ! audioconvert !---->
>
> This is working fine.
>
> To pause src1:
>      unlinking audioconvert src and  sink_0 by gst_pad_unlink and called
> gst_element_release_request_pad  for sink_0.
>      and setting filesrc, decodebin, audioresample, audioconvert to
> GST_STATE_PAUSE
> Src2 is continue to play and its working as expected.
>
> Now after some time wants to resume the src1 while src2 still playing.
> filesrc ! decodebin ! audioresample ! audioconvert ! --->
>                                                                             
> (sink_01) (sink_02) adder ! audioconvert ! autoaudiosink
> filesrc ! decodebin ! audioresample ! audioconvert !---->
>
>          linked audioconvert src and  sink_2 by gst_pad_link and setting
> filesrc, decodebin, audioresample, audioconvert to GST_STATE_PLAYING
>
> In that case Src2 also stopped playing and getting underflow error.
> From gstreamer log its seems that src1 is not pushing any data after setting
> elements to GST_STATE_PLAYING
>
> Any suggestion how to achieve pause/resume operation with adder element. 
>
> Regards,
> Avinash
>

If you pause/unpause branches you might want to try liveadder. Also you
don't describe how you re-link src1. Are you using pad-probes? We have
some examples under gst-plugins-base/tests/examples/dynamic/

Stefan

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You can try to set provide-clock to false on your audio sink.

Which version of GStreamer are you using and on which OS ?

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[avinashgst]

Hi Stefan,

I am able to perform pause/resume successfully with adder after following
modification.

1)  Pause
      -- gst_pad_add_probe (audioconvert src, GST_PAD_PROBE_TYPE_BLOCK |
GST_PAD_PROBE_TYPE_BUFFER,
      (GstPadProbeCallback) cb_have_data, NULL, NULL);
      -- unlinking audioconvert src and  sink_0 by gst_pad_unlink
      -- gst_element_release_request_pad  sink_0

2)  Resume after some time
    -- Linked audioconvert src of Src1 and  sink_2 ( getting by
gst_element_get_request_pad(adder,"sink_%u");) by gst_pad_link
    -- gst_pad_remove_probe (audioconvert src ,probe_id);


Regards,
Avinash



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