> On Dec 12, 2018, at 12:10 PM, Paul B Mahol <[email protected]> wrote: > > On 12/12/18, Ronak <[email protected]> wrote: >> >> >>> 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? >> > > 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. > _______________________________________________ > ffmpeg-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > [email protected] with subject "unsubscribe". _______________________________________________ ffmpeg-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email [email protected] with subject "unsubscribe".
