I don't think that software should reduce the volume upon decoding to an unpredictable level (based on a few short high frequency peaks) unless specifically requested to perform normalization. When playing PCM floating-point format files, Winamp scans the whole fle for peak level, normalizes, and the normalization cannot be defeated. I believe it is incorrect, and a file should be replayed at the level it is recorded by default. (Or it would have been reduced in level.)
It is possible that clipping is audible in some cases, but I haven't ever heard it in music. Tracks subject to the loudness war usually contain enough distortion to effectively mask additional clipping noise. I've heard clipping in recordings of speech, at low bitrate and sample rate, where the clipping is longer in duration, and also due to speech signal being less busy than most music.
Integer overflows sound really nasty. An older version of the WavPack plug-in for Winamp had these (when playing floating-point files, this is now fixed) and also the latest official build of Speex command-line decoder (the Winamp plugin works correctly).
There's an article on medium that Mozilla just posted in their newsletter about this....
Although I believe it's more about a future broadcast
Yep, it works just fine here, Win 10 64 latest version, latest version FB2K, loads of plugins etc. Drop files from Explorer straight into FB2K to create a playlist. It has never not worked (Win 98SE, XP, 7 and now 10).
I always check the Opus 1.2 alpha page to see if new compiled binaries are available and yesterday, when I posted this question, the latest beta was still not posted ^_^
Last post by Chibisteven -
The only thing I wouldn't like is an electric panel that made any bit of an audible hum/buzz. However that might be indicative of electrical problems that need addressing before the fire department has to be called to put out an electrical fire.
I went to a school where this one circuit panel in the hallway was always buzzing. They never fixed the damn thing and to my knowledge is probably still doing it to this day.
I have a UPS for my computer by a subwoofer and I have no problems with either device being by each other. The UPS keeps taking the rumbles just fine and the subwoofer isn't distorted at all. My problem is limited under the desk space.
If you are really paranoid you should measure the analog output of each individual playback device for each song instead of at software level since DACs can introduce intersample peaks as well.
Last post by Arnold B. Krueger -
[Note - I'm not quite sure this is the correct section; mods please move if appropriate]
Why should there be any at all?
We're doing some home upgrades. Unfortunately as part of these upgrades we must move our electrical panel. It currently sits in a stairwell, and that was unfortunately disallowed in a 2008 code revision. Even more unfortunately, the most practical location (i.e. doesn't involve cutting into a century-old plaster wall in a heavy-traffic room) is in the basement. Due to the HVACs vents in our unfinished basement restricting height in most locations to below the 6'6" required by code, placement is further limited. (Code requires a clear space zone that's 6'6" high, 30" wide, and 36" deep.) Long story short, the electrician wants to put the panel in a spot in the basement - right where my subwoofer amp currently sits!
Last post by halb27 -
As far as I understand that this would just mean transforming the input signal from the time domain to the frequency domain. I wouldn't have the hope that this can lead to a bitrate reduction comparable to that of lossyFLAC or wavPack hybrid.
Last post by Arnold B. Krueger -
[Right. So it does not (only) matter what can be reproduced to the LP format, but what a turntable setup can get out of the groove.
To balance out their desire for supersonic hiss they cannot hear and which is only noise anyway, audiophiles must certainly equip themselves with a laser turntable in order to listen to those subsonic signals that never were intended to make it to the medium.
Laser turntable doesn't help because the laser system ends up containing an analogous mechanical system for tracking the disk.
Last post by radorn -
For many lossless codecs, there's LossyWAV with it's variable reduction of bitdepth as an approach to near-losslessness.
For a time now I've been wondering, from a completely inexpert standpoint, about a different approach based on similar concepts as used in lossy codecs.
Instead of aiming to replicate a waveform down to the bits, the idea would be to encode the audible band (or a more limited range, if desired) in the same or similar fashion as lossy encoders do, with the exception that there wouldn't be any psychoacoustic evaluation.
The aim of this would be to reduce bitrate further than lossless and lossywav, but not raising the noisefloor, and also avoiding completely(?) any psychoacoustics-induced artifacts, avobe or under the threshold of audibility, even surviving further processing/DSP and other non-linearities in an audio system that might reveal otherwise inaudible artifacts as present in common lossy compression. For example, on my phone, through headphones I can tolerate very low bitrates without being annoyed by artifacts (or even noticing them), but the same files can become almost unbearably artifacted through the built-in speaker.
I imagine that this could also prove a good target for transcoding of existing lossy material for which there's no access to a better source than old/rare/unsupported formats and neither pure lossy or lossless targets seem appropriate, one introducing further distorsion and the other being a waste of bandwidth (in my limited
Of course, all of this is just hypotetical. All that remains is to answer these questions:
Does something like this exist?
Is it feasible to make an encoder that works like this for an existing lossy codec? Preferably something fully open, such as Opus
Could such an encoding method deliver on the goals of bitrate reduction (compared to lossless), and no artifacting even under harsh conditions (strong non-linearities, heavy use of DSP...)?
Does the concept have any merit or it's just me daydreaming?