[FFmpeg-user] Dynaudnorm & earwax filters

[Ronak]

> On Dec 12, 2018, at 12:10 PM, Paul B Mahol <onemda at gmail.com> wrote:
> 
> On 12/12/18, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>> 
>> 
>>> On Dec 12, 2018, at 11:36 AM, Paul B Mahol <onemda at gmail.com> wrote:
>>> 
>>> On 12/12/18, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>>> 
>>>>> On Dec 12, 2018, at 11:26 AM, Paul B Mahol <onemda at gmail.com> wrote:
>>>>> 
>>>>> On 12/12/18, Ronak <ronak2121-at-yahoo.com at ffmpeg.org> wrote:
>>>>>> 
>>>>>> 
>>>>>>> On Dec 12, 2018, at 8:32 AM, Nicolas George <george at nsup.org> 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?
>> 
> 
> You should send only DBL to this filter.

Sorry I misquoted.

[volume normalization @ 0x7fa4c860dd80] auto-inserting filter 'auto_resampler_0' between the filter 'input' and the filter 'volume normalization'
[auto_resampler_0 @ 0x7fa4c8610940] ch:2 chl:stereo fmt:fltp r:44100Hz -> ch:2 chl:stereo fmt:dblp r:44100Hz

It is being resampled to DBLP.

Besides doing a whole bunch of trial and error, are there any recommended options to use here?

I'm writing one frame of PCM audio into the filter at a time, within my playback audio graph.

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