Last post by j7n -
Another source for general noisiness in loud albums is Clipping. It sounds like a few clicks during loudest monents, identifyable especially on bass or vocals, or a sustained crackle. Spectrally complex sections with multiple instruments sound like a fuzzy noise was mixed with them. Clipping is intentionally used because in many cases it is more transparent than a limiter, but this cannot be generalized to music with soft, pure sounds or human voice.
In waveform view, clipping looks like the top of the waveform was sliced off. It must be evaluated fully zoomed in on multiple loud sections. If the wave is scaled to fit in the range with a limiter, and has normal rounded peaks, then this is not clipping, and cannot be fixed.
In spectral view, clipping looks like a line at the top of the graph for each clipping event. Sustained clipping merges these lines into a band of high-frequency energy, which obstructs features in the original signal.
Unlike compression, clipping can be undone quite sucessfully by extrapolating the signal shape from surrounding data. But the tools for that are not free (free ones aren't very good). The best are Stereo Tool's declipper, and Izotope RX Elements.
For a spectral analyzer I like to use SoX. It is a free command line tool that can generate spectrograms and save them to disk for later viewing. The advantage is that I can scroll through the spectrogram quickly on a slower computer without waiting for it to be rebuilt, while looking a the waveform in another editor. Also, the zoom level can be fixed, which allows to compare spectra of multiple versions of a recording, quickly switching between them. Modern audio editors often have free zoom, and two windows are hard to align. SoX can be added to customizable front-ends, such as Frontah, to avoid using the command line.
Seeing fast amplitude changes on the spectrogram is not easy. To hear an exagerrated effect of a limiter, you could listen to the side/difference channel, subtracting left from right.
Last post by kode54 -
Dosbox resamples it from 49716Hz to whatever rate you have set, using a running average resampler. I resample using a sinc resampler.
It also depends on which Adlib emulator you're using. The "Nuclear Option" drivers use Nuked OPL3, while AdPlug uses the Dosbox "fast" emulator, mainly because it defaults to using 10 instances at once.
That may, or may not, be helpful. It's usually easier to hear something in an audio recording than to see it. And if you haven't done so already, you may be able to zoom-in and see the defect in the regular waveform view.
...And then the issue is, whadda' you gonna' do if you see it?
I recently used MediaMonkey & Mp3Tag to re-rip all my CDs to FLAC, and I'm still hearing TWO (what I perceive to be) different artifacts (in some albums) that I am interpreting as having been introduced in the engineering phase.
Lots of things can go wrong during the production process. Whatever you're hearing, the producer/engineer may not have considered a defect, or maybe it wasn't worthwhile (or possible) to fix or re-record, etc.
...Every time I've though I heard an MP3 compression artifact, it's turned-out that the CD had the same "defect".
I've read the term "pump" being used here and elsewhere and it certainly describes the gawd-awful artifact that I find so irritating,
Dynamic compression (not to be confused with file compression like MP3) is a kind of fast automatic volume control... If something loud comes-along (like a loud kick-drum) the volume is momentarily turned-down and sometimes you can hear the volume "pumping" up and down. Or, there is something called "ducking" where the bass guitar is turned-down momentarily with every drum kick, etc. If it's done right you shouldn't hear the side-effects, but it can be done wrong or over-done.
I think this would show a held chord or note being dropped a split second BEFORE a bass note hits
There is actually something called "look ahead", and if done right it can allow compression/limiting without distortion and perhaps with less pumping). That is a feature of digital processing... The older analog compressors didn't have memory or delay so they couldn't "look ahead" and they can only react to what's happening or what has already happened.
So far I have mainly used high bitrate files, and haven't noticed any problems with either AAC, MP3, MPC or Ogg Vorbis in normal, casual listening. I might do some testing to see if I can find the threshold for transparency for me with different codecs. Or I might not. I mostly deal with high bitrate files anyway.
It seems like there is no particularly "wrong" way to listen via Bluetooth. I was thinking there might be some sort of consensus in favour of AAC and aptX, but I had obviously misunderstood. The whole Bluetooth thing is still a bit cloudy to me, as long as I can't know for sure how different files are handled at all times.
Last post by EpicForever -
"Must have" is VERY subjective term. Reasons can vary - author didn't submitted certain DSP just because he decided so, author simply haven't registered to the components website.
Last post by Studio 308 -
It works as advertised and it will be more accurate now, but why it is still not sorting alphabetically? The drop-down list is sorted (and was always), but context menu is not. What's the point? Conversion presets are sorted everywhere, for example. It is good that you can do anything manually in text file, but it is not good design to do something outside the app.
Last post by H_Allen -
tiny bit of background: I used to listen to MP3's for convenience, but rejected them, and listening to music entirely due to wearying artifacts
I recently used MediaMonkey & Mp3Tag to re-rip all my CDs to FLAC, and I'm still hearing TWO (what I perceive to be) different artifacts (in some albums) that I am interpreting as having been introduced in the engineering phase. One is general "noisiness" -- the worst offender I've found being "The Killers" (any album!), and the other seems to be a selective attenuation of A TRACK in the mix in order to "make room" for another -- usually a bass track.
I have been looking for an audio analyzer that would show music with the x axis being time (as it goes by), the y axis being frequency (high at the front to low at the back) and z being amplitude or volume. I think this would show a held chord or note being dropped a split second BEFORE a bass note hits -- the sort of thing that might be done with a cheap DSP in a purpose-build "Arduino" sort of product.
I've read the term "pump" being used here and elsewhere and it certainly describes the gawd-awful artifact that I find so irritating, but I don't know enough of the vocabulary to identify it in a way others readily understand.
Is there a primer on the common vocabulary used for this stuff? Is there a free or affordable audio analyzer that would use the x-y-z axis presentation I described? Are these artifacts real, or are they in my head? Thanks for your patience!