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Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #100 –
Internally, block size is determined by sample-rate, e.g. 512 samples for 44.1/48 kHz; 1024 samples for 88.2/96kHz; etc..

FLAC blocksize can be varied independently of lossyWAV processing block-size.

Recently I have been using foobar2000 to convert 44.1/16 FLAC to lossyFLAC - setting foobar2000 to output 24-bit as input to lossyWAV (and scaling by 0.5 to give the ANS "room" to work).

--static and --dynamic set the minimum-bits-to-keep for each sample from a fixed datum (--static) or a dynamic datum based on block noise (--dynamic).
lossyWAV -q X -a 4 -s h -A --feedback 2 --limit 15848| FLAC -5 -e -p -b 512 -P=4096 -S-

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #101 –
Quote from: Nick.C on
Internally, block size is determined by sample-rate, e.g. 512 samples for 44.1/48 kHz; 1024 samples for 88.2/96kHz; etc..
Yes I've read about this; I was asking if it's possible to alter this to let LossyWAV use, for example, block size of 4096 samples for 44.1 kHz.

some ANC'd headphones + AutoEq-based impulse + Meier Crossfeed (30%)

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #102 –
Would this also work for MP3's? 
I ask because VBR MP3's could do either nicer or smaller if the sample rate (bitdepth) is reduced/optimized before encoding. 

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #103 –
It won't work for MP3.

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #104 –
Quote from: lvqcl on
It won't work for MP3.
Indeed.

@Max9000The method used by lossyWAV doesn't work with MP3 - and lossyWAV isn't competitive in terms of bitrate with MP3 either.
lossyWAV -q X -a 4 -s h -A --feedback 2 --limit 15848| FLAC -5 -e -p -b 512 -P=4096 -S-

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #105 –
Hi Nick.  And thanks!

I asked because the barrier to good small mp3 files was CD 44100 sample-rate spec (harms/exhausts small-rate mp3's); but, high quality small files can be had with 32000 sample rate, so long as it wasn't done with the mp3 codec's inbuilt fast-n-dirty resampler.   Foobar's PPHS set to Ultra will do much better than the codec's inbuilt resampler.

The trouble with non-adaptive sample rate reduction is noise floor (like a slower LP or tape). 

So, I'm curious about PPHS-ultra@32000 > LossyWAV > WavPack lossy@460 (also quality max) > GXlame -V30.

Basically, if a LAME variant can't toss the extra bits, then an interstage of WavPack might get it.  I'm just not sure if I got the chain right.  Where is the best spot for the 32000 conversion? 

P.S.  Application is internet radio. 
Could be really useful if:  Archive quality at smaller file size > also results in smaller production bandwidth consumption.  Is that feasible? 

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #106 –
Quote from: Max9000 on
The trouble with non-adaptive sample rate reduction is noise floor (like a slower LP or tape). 

I don't fully understand your idea, but LossyWAV increases noise floor, not decreases it. So this --
Quote from: Max9000 on
So, I'm curious about PPHS-ultra@32000 > LossyWAV > WavPack lossy@460 (also quality max) > GXlame -V30.
-- makes zero sense.

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #107 –
It would make more sense if I could use LossyWAV as an opportunistic resampler 44100 (or any) input, to 32000 (or lower) output. 
That would actually make smaller WAV files which could be used for smaller AAC, smaller MP3, etc...

Perhaps it is a prospective cool new feature for LossyWAV 1.5.0?

Re: lossyWAV 1.4.2 Development (was 1.5.0)

Reply #108 –
Quote from: Max9000 on
The trouble with non-adaptive sample rate reduction is noise floor
Aren't you confusing downsampling with bit depth reduction?
 
 
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