[FFmpeg-user] Dynaudnorm & earwax filters

[Paul B Mahol]

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.