[FFmpeg-devel] [PATCH] libswresample: avoid s16p internal processing format
Paul B Mahol
onemda at gmail.com
Thu Jan 12 18:09:18 EET 2023
On 1/12/23, Michael Niedermayer <michael at niedermayer.cc> wrote:
> On Thu, Jan 12, 2023 at 03:20:06PM +0100, Paul B Mahol wrote:
>> On 1/8/23, Michael Niedermayer <michael at niedermayer.cc> wrote:
>> > On Fri, Jan 06, 2023 at 07:04:59PM +0100, Paul B Mahol wrote:
>> >> On Fri, Jan 6, 2023 at 7:01 PM Paul B Mahol <onemda at gmail.com> wrote:
>> >>
>> >> >
>> >> >
>> >> > On Fri, Jan 6, 2023 at 6:25 PM Michael Niedermayer
>> >> > <michael at niedermayer.cc>
>> >> > wrote:
>> >> >
>> >> >> On Thu, Jan 05, 2023 at 11:08:25PM +0100, Paul B Mahol wrote:
>> >> >> > On Thu, Jan 5, 2023 at 9:53 PM Michael Niedermayer <
>> >> >> michael at niedermayer.cc>
>> >> >> > wrote:
>> >> >> >
>> >> >> > > On Thu, Jan 05, 2023 at 01:44:10PM +0100, Paul B Mahol wrote:
>> >> >> > > > Patch attached.
>> >> >> > >
>> >> >> > > > swresample.c | 3 ++-
>> >> >> > > > 1 file changed, 2 insertions(+), 1 deletion(-)
>> >> >> > > > eee7a0685b44aa867562138a2e2437ecb8844612
>> >> >> > > 0001-libswresample-swresample-avoid-s16p-internal-transfe.patch
>> >> >> > > > From 9c4cd60e2dd41cf98d693c8251f4cfade0807073 Mon Sep 17
>> >> >> > > > 00:00:00
>> >> >> 2001
>> >> >> > > > From: Paul B Mahol <onemda at gmail.com>
>> >> >> > > > Date: Thu, 5 Jan 2023 13:40:12 +0100
>> >> >> > > > Subject: [PATCH] libswresample/swresample: avoid s16p
>> >> >> > > > internal
>> >> >> transfer
>> >> >> > > format
>> >> >> > > >
>> >> >> > > > Instead use float one by default for sample rate conversions.
>> >> >> > > > The s16p internal transfer format produces visible and
>> >> >> > > > hearable
>> >> >> > > > quantization artifacts.
>> >> >> > >
>> >> >> > > When does this occur and why?
>> >> >> > >
>> >> >> >
>> >> >> > It occurs always. Just compare output with 16bit and
>> >> >> > int32/float/double.
>> >> >> > Look at other people report on internet.
>> >> >> > Look at src.infinitewave.ca
>> >> >>
>> >> >> src.infinitewave.ca uses 32bit none of what it shows should touch
>> >> >> the
>> >> >> codepath
>> >> >> you change.
>> >> >>
>> >> >> if we look at src.infinitewave.ca for swr we see 2 types of
>> >> >> artifacts
>> >> >> 1. Aliassing which is at maybe -120db with the actual signal at 0db
>> >> >> i would like to see some evidence that a human can hear this
>> >> >>
>> >> >
>> >> > For s16p<->s16p it is much lower, around -78dB thus this patch.
>> >> >
>> >> > Also for others and reports for swr its is lower than exact -120dB
>> >> >
>> >> >
>> >> > 2. Reflection and attenuation at the transition frequency
>> >> >> With linear filters there is a tradeof between attenuation of the
>> >> >> passband, reflection of frequencies beyond, latency and so on
>> >> >> You can have a perfect sharp cutoff with no attenuation and no
>> >> >> refelection
>> >> >> that requires a infinitly long filter. And while this looks best in
>> >> >> this
>> >> >> frequency plot, does it actually sound best ? If you can hear
>> >> >> -120db
>> >> >> signals you surely would then also hear the ringing long before a
>> >> >> gunshot
>> >> >> from such long filter.
>> >> >>
>> >> >
>> >> One can always change linear FIR to be min phase FIR kernel.
>> >
>> > I certainly would welcome a wider range of filters in swr, if you want
>> > to
>> > add
>> > any low delay sinc approximation or in fact i would welcome any filter
>> > you want to add.
>>
>> There is that afdelaysrc filter patch on ML to add FIR coefficient
>> generation for fractional delay audio filter that can be also used as
>> a interpolation FIR filter. And to me it seems better at same number
>> of taps than already used/available ones in soxr and swr.
>
> Please add improvments into swr, if you have any!
>
>
>>
>> Also I have done prototype of resampling filter using afir filter via
>> custom filters in filtergraph and it operates at similar speeds like
>> soxr (in these very non optimized approach) and providing better/wider
>> frequency output at highest band.
>
> Can it be added into swr ?
It will use libavutil/tx.h for convolution.
But I need first to research more and write actual code that does
out/in sample rate ratio factorization and do lot of benchmarks
comparing normal ratios with very big ones and make sure that approach
is always faster than swr/soxr and at same time providing better/same
quality.
Also need to add min phase version of filter and compare what
performance/latency it can bring at all.
>
> thx
>
> [...]
>
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> Awnsering whenever a program halts or runs forever is
> On a turing machine, in general impossible (turings halting problem).
> On any real computer, always possible as a real computer has a finite
> number
> of states N, and will either halt in less than N cycles or never halt.
>
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