On 12/12/18, Ronak <[email protected]> wrote: > > >> 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. >
I can not guess, need to look at source code. _______________________________________________ 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".
