On fly sound latency change

[Henrikas Urbonas]

Dynamic latency changes feature to jitter buffer would not solve chopped
sound issue if we will be reducing latency on fly.
Then there should additional plug in that would seamlessly adjust buffer
time when we dynamically change sound latency from higher to lower real
time.

Regards gstreamer on Android we see fixes proposed to gstreamer in
sourcecode however none of them were implemented. Because on our previous
projects there were no problems with lower output latency with 48 kHz
sample rate.

2015-03-20 18:40 GMT+02:00 Stefan Sauer <ensonic at hora-obscura.de>:

>  On 03/20/2015 05:04 PM, Henrikas Urbonas wrote:
>
> When you change "latency" while pipeline is running, nothing happens.
> From documentation it looks like latency can be set and have effect only
> before pipeline is in playing state.
> Because if you change latency when pipeline is in playing state it
> actually does not change the latency (we use jitter buffer).
>
> since you've tested this, the next step would be to file a feature request
> for the jitter buffer to allow dynamic latency changes.
>
>
>  Are there other ways to change latency except using rtspsrc jitter
> buffer?
>
> This is at least one of the elements that in the pipeline that introduce
> latency and that can be configured. For other elements the latency might be
> static.
>
>  And why Android cannot use 48 kHz sample rate because some devices have
> lower output latency when using this (48kHz) sample rate?
>
> I am not using gstreamer on android. You would need to start looking at
> the audio source and sink elements and figure out why they won't work at
> 48kHz (if that is the case).
>
> Stefan
>
>
>
>  Thanks guys
>
>
> 2015-03-20 10:12 GMT+02:00 Stefan Sauer <ensonic at hora-obscura.de>:
>
>>  On 03/18/2015 05:33 PM, Henrikas Urbonas wrote:
>>
>> So that means to adjust it without glithces we have to write a plugin for
>> Gstreamer?
>>
>>
>>  I think most part of the latency is the buffering time (e.g. in the
>> jitter buffer). I would first try what happens if you change the 'latency'
>> parameter while the pipeline is running.
>>
>> STefan
>>
>>
>> 2015-03-18 18:23 GMT+02:00 Tim Müller <tim at centricular.com>:
>>
>>> On Wed, 2015-03-18 at 17:54 +0200, Henrikas Urbonas wrote:
>>>
>>> Hi,
>>>
>>> > We work on RTSP protocol and we want to dynamically change sound
>>> > latency on client side without re-initializing the pipeline, did
>>> > anyone tried it?
>>>
>>> I think changing the latency on the fly at runtime should work fine
>>> already, but what we don't have yet is a way to smoothly adjust the
>>> latency over a period of time so that you can do adjustments without
>>> glitches.
>>>
>>>  Cheers
>>>   -Tim
>>>
>>> --
>>> Tim Müller, Centricular Ltd - http://www.centricular.com
>>>
>>> _______________________________________________
>>> gstreamer-android mailing list
>>> gstreamer-android at lists.freedesktop.org
>>> http://lists.freedesktop.org/mailman/listinfo/gstreamer-android
>>>
>>
>>
>>
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>
>
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