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PostPosted: Wed Jul 21, 2021 2:18 pm 
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Joined: Wed Nov 08, 2017 3:16 pm
Posts: 141
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Question about input, why when input2 is ON, no matter what Asio or no Asio 48kHz,, 192.. buffer on input1 will start to overfill after some time?
https://i.imgur.com/wivmKwZ.png
that is generally indicative of a sample timing issue, i have this same issue as well but using ST output to HDMI audio device.
buffer slowly filling means device sample rate is slightly lower than ST sample rate (ex. 47,999 > 48,000).
buffer slowly emptying means device sample rate is slightly higher than ST sample rate (ex. 48,001 > 48,000).
if there was a way to fine-tune the sample rate on ST to match that of input/output devices, this issue could be resolved.


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PostPosted: Wed Jul 21, 2021 2:49 pm 
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Joined: Mon Mar 17, 2008 1:40 am
Posts: 10339
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Question about input, why when input2 is ON, no matter what Asio or no Asio 48kHz,, 192.. buffer on input1 will start to overfill after some time?
https://i.imgur.com/wivmKwZ.png
that is generally indicative of a sample timing issue, i have this same issue as well but using ST output to HDMI audio device.
buffer slowly filling means device sample rate is slightly lower than ST sample rate (ex. 47,999 > 48,000).
buffer slowly emptying means device sample rate is slightly higher than ST sample rate (ex. 48,001 > 48,000).
if there was a way to fine-tune the sample rate on ST to match that of input/output devices, this issue could be resolved.
That fine-tuning happens automatically on all inputs and outputs EXCEPT for input 2. So you shouldn't have problems with this unless you use input 2.


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PostPosted: Wed Jul 21, 2021 3:07 pm 
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Joined: Tue Mar 17, 2009 2:56 pm
Posts: 3797
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Quote:
Quote:
Question about input, why when input2 is ON, no matter what Asio or no Asio 48kHz,, 192.. buffer on input1 will start to overfill after some time?
https://i.imgur.com/wivmKwZ.png
that is generally indicative of a sample timing issue, i have this same issue as well but using ST output to HDMI audio device.
buffer slowly filling means device sample rate is slightly lower than ST sample rate (ex. 47,999 > 48,000).
buffer slowly emptying means device sample rate is slightly higher than ST sample rate (ex. 48,001 > 48,000).
if there was a way to fine-tune the sample rate on ST to match that of input/output devices, this issue could be resolved.
That fine-tuning happens automatically on all inputs and outputs EXCEPT for input 2. So you shouldn't have problems with this unless you use input 2.
So another feature that exists but nobody cant really use it.

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PostPosted: Wed Jul 21, 2021 3:18 pm 
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Joined: Tue Aug 26, 2014 4:52 pm
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Location: Nijmegen
Beta 028 shows 027 stand alone 64bit


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PostPosted: Wed Jul 21, 2021 3:24 pm 
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Joined: Wed Nov 08, 2017 3:16 pm
Posts: 141
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Quote:
Question about input, why when input2 is ON, no matter what Asio or no Asio 48kHz,, 192.. buffer on input1 will start to overfill after some time?
https://i.imgur.com/wivmKwZ.png
that is generally indicative of a sample timing issue, i have this same issue as well but using ST output to HDMI audio device.
buffer slowly filling means device sample rate is slightly lower than ST sample rate (ex. 47,999 > 48,000).
buffer slowly emptying means device sample rate is slightly higher than ST sample rate (ex. 48,001 > 48,000).
if there was a way to fine-tune the sample rate on ST to match that of input/output devices, this issue could be resolved.
That fine-tuning happens automatically on all inputs and outputs EXCEPT for input 2. So you shouldn't have problems with this unless you use input 2.
unfortunately, i am using the first input and have this timing issue on the normal output to an audio device connected via HDMI. i know its a timing issue with HDMI and despite ST attempting to automatically fine-tune, i still have a buffer overflow after roughly 10 minutes of use. the entire chain of audio is digital at 48Khz and 16Bit for ST, input and output as well as the audio device on HDMI audio device which accepts 48Khz, 16bit.
this is HDMI from a PC.
normal output buffer is 30mS.
input block size is 2048.


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PostPosted: Wed Jul 21, 2021 4:07 pm 
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Joined: Mon Mar 17, 2008 1:40 am
Posts: 10339
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that is generally indicative of a sample timing issue, i have this same issue as well but using ST output to HDMI audio device.
buffer slowly filling means device sample rate is slightly lower than ST sample rate (ex. 47,999 > 48,000).
buffer slowly emptying means device sample rate is slightly higher than ST sample rate (ex. 48,001 > 48,000).
if there was a way to fine-tune the sample rate on ST to match that of input/output devices, this issue could be resolved.
That fine-tuning happens automatically on all inputs and outputs EXCEPT for input 2. So you shouldn't have problems with this unless you use input 2.
unfortunately, i am using the first input and have this timing issue on the normal output to an audio device connected via HDMI. i know its a timing issue with HDMI and despite ST attempting to automatically fine-tune, i still have a buffer overflow after roughly 10 minutes of use. the entire chain of audio is digital at 48Khz and 16Bit for ST, input and output as well as the audio device on HDMI audio device which accepts 48Khz, 16bit.
this is HDMI from a PC.
normal output buffer is 30mS.
input block size is 2048.
Ah. If you set the buffer really low, the automatic correction is turned off because Stereo Tool has to assume that you know what you're doing. You can force it to try to correct anyway, by setting "Synchronize to output" in the input settings from Auto to On (or Yes, not really sure how I named it).


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PostPosted: Wed Jul 21, 2021 4:41 pm 
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Joined: Wed Nov 08, 2017 3:16 pm
Posts: 141
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That fine-tuning happens automatically on all inputs and outputs EXCEPT for input 2. So you shouldn't have problems with this unless you use input 2.
unfortunately, i am using the first input and have this timing issue on the normal output to an audio device connected via HDMI. i know its a timing issue with HDMI and despite ST attempting to automatically fine-tune, i still have a buffer overflow after roughly 10 minutes of use. the entire chain of audio is digital at 48Khz and 16Bit for ST, input and output as well as the audio device on HDMI audio device which accepts 48Khz, 16bit.
this is HDMI from a PC.
normal output buffer is 30mS.
input block size is 2048.
Ah. If you set the buffer really low, the automatic correction is turned off because Stereo Tool has to assume that you know what you're doing. You can force it to try to correct anyway, by setting "Synchronize to output" in the input settings from Auto to On (or Yes, not really sure how I named it).
ah yes, thanks for that, i see its about a 0.1hz off from timing (0.1Hz to better describe the de-sync rate).
this corrects my issue, thanks.
the naming of that module threw me off, didnt realize it was sample correcting for output to match the device sample, ahh, the difference between technical terms and laymans terms.
0.01 @ 32, setting corrects my issue completely :)


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PostPosted: Wed Jul 21, 2021 6:26 pm 
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Joined: Wed Nov 08, 2017 3:16 pm
Posts: 141
i know this isnt the correct place to make note of this but:
lowpass filter > filter before processing > enabled
does not work as indicated.
if i generate a 10Khz sinewave tone on input and then move lowpass slider to 9Khz or below, the 10Khz sinewave tone is still passed into processing, its visible in AGC and MB.
it appears the lowpass filter is pre-processing only and highpass filter is post-processing only, despite "filter before processing" enabled or not, "filter before processing" doesnt appear to do anything at all to lowpass and highpass filters.


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PostPosted: Wed Jul 21, 2021 6:36 pm 
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Joined: Tue Mar 17, 2009 2:56 pm
Posts: 3797
Quote:
i know this isnt the correct place to make note of this but:
lowpass filter > filter before processing > enabled
does not work as indicated.
if i generate a 10Khz sinewave tone on input and then move lowpass slider to 9Khz or below, the 10Khz sinewave tone is still passed into processing, its visible in AGC and MB.
it appears the lowpass filter is pre-processing only and highpass filter is post-processing only, despite "filter before processing" enabled or not, "filter before processing" doesnt appear to do anything at all to lowpass and highpass filters.
That filter does not go under 16kHz.

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PostPosted: Wed Jul 21, 2021 7:01 pm 
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Joined: Wed Nov 08, 2017 3:16 pm
Posts: 141
Quote:
Quote:
i know this isnt the correct place to make note of this but:
lowpass filter > filter before processing > enabled
does not work as indicated.
if i generate a 10Khz sinewave tone on input and then move lowpass slider to 9Khz or below, the 10Khz sinewave tone is still passed into processing, its visible in AGC and MB.
it appears the lowpass filter is pre-processing only and highpass filter is post-processing only, despite "filter before processing" enabled or not, "filter before processing" doesnt appear to do anything at all to lowpass and highpass filters.
That filter does not go under 16kHz.
yeah, i get it now, but it would be nice to have this filter selectable for pre or post processing.
i have some live audio content (vocal) that has a constant sinewave tone of 10Khz i would like to eliminate before processing using this filter set at 9Khz (9.9Khz) but its a no-go to and as a result, the constant tone affects the AGC.


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