Last post by Porcus -
Not to mention that there is no way that response at 20 Hz and below can be accurately reproduced by a LP.But maybe by an EP played back at 16 2/3. A switch every kid who had access to it just had to try, right?
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.
Scientific Discussion / Re: A few ideas and questions about audio mediums that have been bugging me.Last post by Porcus -
Actually ... I did not even know about this: https://en.wikipedia.org/wiki/Hi-MD had a capacity for 94 minutes PCM. Brought to you by the creators of Betamax ...
But I cannot imagine industry support - rather, one would have to expect major costs fighting the RIAA back then when they tried to restrict audio-on-CD-R to very special hardware and media.
"considered a mistake that contributed to the Dreamcast's early demise."
Yeah ... so even in the game console world, where the drives are bundled with the machine (unlike for music) and the vendor ensures that this is how the content is delivered (unlike for music), this choice of format could ruin your business.
Last post by halb27 -
This morning I also tested harp40_1 using --bitrate 160 (172 kbps on average for my test set of varios pop music) and --bitrate 192 (205 kbps for my test set) using buildA and build C. I can ABX each of the encoding results.
Using --bitrate 160 I'd call the results acceptable, using --bitrate 192 the issue is negligible to me.
Considering the fact that --bitrate 128 is transparent for nearly everything, --bitrate 160 is my sweet spot giving some (but not exaggerated) headroom for very evil samples.
Ignoring very evil samples --bitrate 96 is great and --bitrate 128 is pefect (judging from what we know so far).
Yep. It sounded exactly like when you apply nearest neighbor algo on samples. Harmonics everywhere! No filtering, nothing!
Not trying to be a smart aleck or something (please--the least thing I want is an enemy) but I am also a real fan of chiptune music. (The fact that Opus does best when it comes to chiptune music as per my ABX tests is another topic but https://hydrogenaud.io/index.php/topic,105808.new.html )
Some computers do apply nearest neighbor resampling on samples, most noticeably the Amiga's sound chip nicknamed "Paula" does this on the hardware level. The Gameboy has a single sound channel that can playback samples resampled using--yep, nearest neighbor. The C64 and Atari XL/XE can also play samples through software mixing, and when it comes to their slow CPUs burdened with realtime audio mixing, we have no choice but to go with nearest neighbor resampling. We can't afford even linear resampling with these computers. A notable exception however, is with the Super NES / SNES / Super Famicom, which uses gaussian resampling on the hardware level (not quite sure if it is indeed called gaussian but I'm certain it doesn't use nearest neighbor or linear resampling--it uses something more advanced than linear but less advanced than cubic.)
Since you mentioned PSG, then that points us to (mostly) non-sample based chips, like the Nintendo's 2A03 or the 3-channel AY chips, or even FM synth ones like the OPL series, maybe even the Gameboy's sound chip. These chips make variable pulse width square waves, and it won't even matter if we resample them using nearest neighbor or cubic--those are square waves The NES (not Super NES) also creates triangle waveforms by playing back an internally stored sample of a (weirdly distorted) triangle wave at various frequencies resampled--yep, nearest neighbor The VRC6 sound chip made by Konami uses an internal counter that increments in fixed time intervals before looping back to zero and incrementing again to create a sawtooth waveform. Analyzing the waveform, it exibits a "staircase"-like wave, produced because of the integer counter which jumps incrementally, the jump size depending on the requested frequency. Since it isn't a perfect sawtooth wave, nearest neighbor resampling wouldn't hurt too much Yet another NES sound expansion chip, the Namco N106 (not sure about the name) is a sample-based audio chip that resamples in nearest neighbor.
My point is that chiptune already uses lots and lots of linear neighbor interpolation / resampling. In my opinion, THAT gives them that unique chiptune feel. Because of that, they already contain large amounts of harmonics (can't blame square waves--that's normal and expected from them) and if a lossy audio codec can't retain those harmonics, then that's not a good codec.
Just my two cents. Please feel free to ignore this or what.
I am not as technical as most people here in the forum, but here's how I consider the 48kHz fixed sampling rate or Opus.
I sometimes find it bad since sometimes, I want to ABX Opus with another codec, but just because Opus outputs 48kHz audio, I have to go for extra measures just to resample the output of the other codec to 48kHz also.
However, honestly, I don't mind Opus' 48kHz output. True, CDs are at 44.1kHz, but besides CDs, many things are operating at 48kHz, like DVDs and Bluray audio. Also, as resampling to a higher sample rate is (at least audibly) lossless (https://www.youtube.com/watch?v=cIQ9IXSUzuM), then I see no problem encoding CD audio in Opus.
I'm the kind of guy who keeps a large number of songs in his phone, so I use low bitrates of 64kbps. Compared to encoding 44.1kHz CD audio using 44.1kHz AAC-HE where I can notice bad low bitrate artifacts, I would rather use 48kHz Opus and appreciate the fact that this is the best 64kbps audio can get.
I'm all for supporting Opus, no matter if it suddenly decides to output 24-bit 48kHz audio, as long as it improves the audible sound quality.
Not looking for an argument here--just stating my own side.
I want to be able to build opusenc and libopus from source using Visual Studio 2017, or if needed, Code::Blocks.
While I am indeed knowledgeable in programming and compiling stuff, I'm just completely clueless when it comes to compiling other people's code.
I know that building this in Linux would be way easier (and indeed, I have successfully built both under Linux LOL ) but I really need to be in Windows this time.
Is there some website I can read that can help me, or maybe someone can give me instructions, possibly?
I no longer want to wait for others to build opusenc every time a new version comes out XD
Last post by IgorC -
Thank You, Halb
I've tried harp40_1
Agree, 96 kbps wasn't enough for acceptable quality. 128 kbps was easy abxable but differences weren't such annoying as at 96 kbps.
I just released 1.2-beta, incorporating this change. To be more precise, the change I made in build C is fully enabled below 64 kb/s and then gradually phased out between 64 kb/s and 80 kb/s. For now I didn't want to change the behaviour at high bit-rate just to be on the safe side (it's already high quality so I want to make sure I don't make anything worse).Yes, it's understandable. Glad to see Kamedo2's and my results were useful.
Opus hits 1.2-beta. Nice.
Also I think build C demonstrates a potential of possible smart adaptive bitrate distribution for LF/HF (in future?). Anyway build C already does good job here and You guys work hard on AV1 video format so I don't pretend to see drastic quality changes those will require a huge amount of work. (hm, neither I have submited any useful tests in previous months)
Scientific Discussion / Re: A few ideas and questions about audio mediums that have been bugging me.Last post by Phanton_13 -
Any link to any tools and documentation for such thing? In theory alot can be possible but I haven't been able to find solid proof for such thing, like some kind of command line app that modulates the audio.
The documentation is rare, and the software nonexistent as all the software is tailored to compatibility with old formats or based on it.
This paper describes a technique that clearly have much similarity with the one used in DCC:
One paper about DCC:
This paper is interesting and relates to posterior developments and more in your line.
We also have the streamer cassette that hold up to 160MB but the information is very scarce.
And for software I don't know any but you can try to use DRM software to do a proof of concept, DRM as in Digital Radion Mondiale. Although is more tailored to radio transmission it can work quite well in tape as the radio medium is more noisy.
In my case, I changed from EVENT mode to PUSH mode and it started working again. I'm never really sure what the different is between these modes but usually one of them will work
Hi - new Win10 PC running latest Creators Update, v1703. Using the new native Windows 10 USB audio drivers, selecting WASAPI Event Style, music plays though my USB DAC and sounds fine but the progress bar does not move.
Yes, same problem. Also the elapsed time doesn't update, and the frequency range visualization doesn't change. I reported this under another post (Playback doesn't follow cursor, and now the reverse). It's REALLY difficult to use foobar when I can't stop and reset to a certain point, etc. Still haven't seen any responses.
Hmmm... changed Event to Push mode and it works as before....