Did you read the hydrognaudio thread carefully? The post was talking about the original command line version of SoX has clipping problem, and the foobar plugin version does not clip since it uses float. Exactly one of the advantages of floating point processing.
Sorry, it is you who misunderstood that hydrogenaudio thread, and if you read carefully I am also the OP of that thread.You're completely misunderstanding the Hydrogenaudio thread. There is no clipping problem in the SoX command line version. Any resampling to higher sample rates risks creating inter-sample peaks above FS; the command line version of SoX notes this, warns you, and tells you to add headroom so it doesn't happen. The reason this guy added float support to Foobar's SoX plugin is because he wants to pass above FS floats down through Foobar's pipeline to potentially be dealt with later on with Foobar's normal limiting.
I just realized that you're the author of the guest piece on Archimago's blog. I think you'd probably benefit from some reading on basic DSP concepts so you understand what is going on.
Some MPC-HC tests:-MP-HC (Media Player Home cinema)?
If I recall correctly, the documentation of MPC-HC player does not recommend setting it to max volume, but to 75-80 % for best results.Thank you !
Volume MPC-HC strongly improve if you set max volume at 95% : strange but measurements give the demonstration !
You may have a look at this article:
http://archimago.blogspot.com/2019/06/guest-post-why-we-should-use-software.html
For example, with an ideal 24-bit device, it is possible to playback a 16-bit file 48dB lower without losing quality, because one bit has about 6dB of dynamic range (the exact formula of bitdepth and dynamic range is 6.02*n-bits + 1.76).
Yes, but the point is if the original file is 16-bit then a properly dithered > 16-bit digital volume control won't degrade the signal until the bit-depth limit is reached. Also, it is not the main point of the article, please read until the end.Music isn't linear - some frequencies are louder than others. Here is an example. It's a spectrum of a pop song. As you can see, 50Hz is the loudest frequency: its amplitude is -5 dB. 10KHz is -40dB quieter than that. For a 16bit recording this means dynamic range of the 50Hz sounds is 15bits and just 9bits for 10KHz. We're already at the point where amplitude resolution isn't that great.
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It is because those classic WAVES plugins work with hardware working in fixed point like Pro Tools TDM. The newer generation of plugins work in 64-bit float already.
As you can see, 50Hz is the loudest frequency: its amplitude is -5 dB. 10KHz is -40dB quieter than that. For a 16bit recording this means dynamic range of the 50Hz sounds is 15bits and just 9bits for 10KHz. We're already at the point where amplitude resolution isn't that great.
Interesting. I am no longer using those Waves plugins as there are a lot of other choices these days. If you did nothing other than bit conversion then harmonics at -140dB is still pretty high even for 32-bit float. For instance here is the 15 years old Adobe Audition 1.5 generating a sine then dithered to 24-bit (white) , and the free Audacity (blue) doing the same thing, analyzed by DeltaWave:Feeding a clean 1kHz sine (-1dBFS) into L1+ (threshold at 0dBFS), with Dither set to "24-bit, Type 1, Normal shaping," added harmonics can be seen in a spectrum analyser; they are >140dB down, but they are there--and, by way of comparison, using the JS* plug-in "Bit Reduction and Dither with Psychoacoustic Noise Shaping," only the noise floor is visible.
Interesting. I am no longer using those Waves plugins as there are a lot of other choices these days. If you did nothing other than bit conversion then harmonics at -140dB is still pretty high even for 32-bit float.
The height of the harmonics are related to sample rate and the frequency of the test tone. For example, try to generate a 6kHz sine at 48kHz, then try 5999Hz.Using the same Reaper "JSFX" Spectrum Analyzer with identical settings (albeit 96kHz sample rate, so relatively shorter FFT size)...
Reaper "JSFX" Tone Generator -- 1kHz sine (-1dBFS amplitude):
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Reaper Export to 32-bit FP:
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L1+ Dither to 24-bit:
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Reaper "JSFX" Bit Reduction and Dither w/ Noise Shaping:
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Harmonic distortion components are far below -140dB on these plots, but they are clearly visible.
The height of the harmonics are related to sample rate and the frequency of the test tone. For example, try to generate a 6kHz sine at 48kHz, then try 5999Hz.
In real (even synthesized) music, the spectrum is rarely a single tone and the truncation effect is negligible, and the practical benefit for floating point is clipping prevention rather than the inaudible low level distortion.
In fact there is another (locked) thread about this issue:The other aspect was to show that one should not make any assumptions about the behaviour of software DSP processes. In this case, the artifacts may well be benign. However, I have measured "professional" plug-ins for which the cost of a license is into three figures, and yet garbage appears at the output, such as spurious tones around 10kHz.
What (modern) "PC" software uses fixed-point DSP for audio processing?