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General Audio / Re: Another audiophile "expert"
Last post by forart.eu -Well, since people still use SoX, and its 32bit int processing domain, nothing surprises me any more.Agree: 64fp would be better.
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FLAC / Re: FLAC v1.4.x Performance Tests
Last post by ktf -3
Speech Codecs / Re: No ACELP.KELVIN encoder!
Last post by Octocontrabass -You should code a encoder from PCM WAV to ACELP.KELVIN because that's like G.729 but also supports stereo and compile to EXE that works with Windows 7. Post also the source code (as a ZIP file).G.729 already supports stereo. You don't need acelp.kelvin for that.
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3rd Party Plugins - (fb2k) / Re: Dynamic Range plugin
Last post by Tropoje -Quote
Log option added. I did my best to mimic the formats. The log uses different syntax for single files.Thank you
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FLAC / Re: FLAC v1.4.x Performance Tests
Last post by Replica9000 -Wav: 24-bit/48 KHz 2h33m 2.47GiB
CPU: Ryzen 5850U
flac git-04532802 (2024-05-02)
Code: [Select]
1 thread 4 threads 8 threads
-5 0m14.505s 0m4.750s 0m4.733s
-5p 0m34.664s 0m10.664s 0m7.061s
-8 0m54.564s 0m16.920s 0m10.359s
-8p 6m13.161s 1m57.957s 1m14.785s
flac git-1ab3c8e7 (2024-05-15)
"Improve calculation of when to use wide residual computation. This change should make 24-bit encoding faster, because the limit_residual variant of residual computation is used less often"
Code: [Select]
1 thread 4 threads 8 threads
-5 0m13.705s 0m4.586s 0m4.719s
-5p 0m24.876s 0m7.702s 0m5.618s
-8 0m42.542s 0m13.355s 0m8.253s
-8p 3m49.137s 1m14.035s 0m47.345s
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FLAC / Re: FLAC-git Releases (Code Base v1.4.x)
Last post by Replica9000 -Linux static binaries
- Built with GCC 12.3.0
- i386, amd64, armv7 & armv8
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Scientific Discussion / Re: Are complex-input FFTs really useful for audio analysis?
Last post by saratoga -BTW, what I mean by "complex-input FFTs" is treating stereo signal as complex numbers (real part = left channel, imaginary part = right channel) and fed into FFT instead of doing two FFTs for each stereo pair
Since the real and imaginary parts of the transform are orthogonal, this is the same as performing a real-valued FFT once for each input and then combining the individual outputs.
See this derivation for a more detailed explanation: http://www.hyperdynelabs.com/dspdude/papers/COMPUTING%20THE%20FFT%20OF%20TWO%20REAL%20SIGNALS%20USING%20A%20SINGLE%20FFT.pdf
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Scientific Discussion / Are complex-input FFTs really useful for audio analysis?
Last post by TF3RDL -BTW, what I mean by "complex-input FFTs" is treating stereo signal as complex numbers (real part = left channel, imaginary part = right channel) and fed into FFT instead of doing two FFTs for each stereo pair
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3rd Party Plugins - (fb2k) / Re: JScript Panel
Last post by marc2k3 -Support for fb2k 1.6. and 2.0 has been dropped. 2.1 is the new minimum. Also windows 10 or later is required.
https://github.com/jscript-panel/release/releases/tag/latest
Old versions remain available:
https://github.com/jscript-panel/release/releases/tag/legacy
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3rd Party Plugins - (fb2k) / Re: foo_truepeak True Peak Scanner
Last post by Defender -New version uploaded: https://foobar.hyv.fi/?view=foo_truepeak
Added extra abort checks. And changed the behavior of info removal. Now if True Peak Scanner is configured to scan and write standard ReplayGain tags the removal will remove those also.
A hint for @Defender: you can hide unwanted context menu entries in Preferences and get them visible by holding shift when you open context menu. That, I think, is a good way to hide seldom used stuff but still have access to it.
Thx. All ok. And thanks for the shift tip.
I noticed you made an option available in the standalone DR scanner to automatically write values without opening the resultscreen.
Can you please implement that option too in truepeak scanner?
SACD scanning is still slow, but I guess it has to do with the extra conversion of the native DSD format.
When I play SACD (I have to use PCM) and I don't use a lowpass filter artifacts are clearly visible in Channel Spectrum. The moment I enable the lowpass filter those artifacts disappear.
I noticed that you always resample SACD to 352k upon scanning which is good. Do you also apply the lowpass before/upon scanning?
If not, wouldn't it be a good idea to do so?
EDIT: Probably best just to activate the lowpass filter in SACD settings itself. That way a conversion to flac does not inherit the artifacts also.