> On Dec 12, 2018, at 11:36 AM, Paul B Mahol <[email protected]> wrote:
> 
> On 12/12/18, Ronak <[email protected]> wrote:
>> 
>>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <[email protected]> wrote:
>>> 
>>> On 12/12/18, Ronak <[email protected]> wrote:
>>>> 
>>>> 
>>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <[email protected]> wrote:
>>>>> 
>>>>> Ronak (2018-12-11):
>>>>>> Ok thanks. I tried to use this filter in my iOS code; but I'm getting
>>>>>> errors with an error code -35.
>>>>>> 
>>>>>> This is my code that tries to write data into the filter graph and
>>>>>> reads it back; what am I doing wrong?
>>>>> 
>>>>> I do not read whatever language that is, but at the very least your code
>>>>> is missing the translation error code -> error message.
>>>>> 
>>>> 
>>>> I found out what my problem is; it's that the dynaudnorm filter is
>>>> returning
>>>> EAGAIN; which means I need to send it more PCM frames.
>>>> 
>>>> Now, I'm trying to integrate this filter into a real time player context;
>>>> and I would like to avoid audio artifacts. I've been playing with various
>>>> options that the filter has; but I can't seem to find one where it would
>>>> work better in the real time context.
>>>> 
>>>> Does anyone know what the correct parameters would be so it works frame
>>>> by
>>>> frame or in a much smaller frame size so we can avoid audio artifacts?
>>>> Alternatively, is there another ffmpeg filter better suited to real time
>>>> dynamic range compression or volume normalization?
>>>> 
>>> 
>>> If you read documentation of filter options you would know.
>> 
>> I already did and tried all sorts of things. I've tried options like:
>> "f=500:g=3:m=10:n=1:b=1", "f=40:g=3:m=10:n=1:b=1". I've even tried the
>> extreme: "f=8000:g=3:m=10:n=1:b=1"
>> 
>> But I still get back lots of EAGAIN.
> 
> That's normal, if you insist on 0 latency look at something else.
> Other players like mpv, handle it fine.

Ok. One last thing is it seems like the filter is spitting out lots of pops and 
crackles when I can get it to return audio frames back out.

Do you know why that would be? I changed all my arguments to just be f="1000" 
since I thought my options would be causing this. But it's not.

Just in case it helps, I am sending in FLTP which is being resampled by the 
rwresample filter to S32. I don't think that would be a factor in this right?

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