all my comments are about ABX so stop trying to be smart with your semantic disingenuousness.
you need to inform yourself about perceptual testing - I won't waste the time.
All I will say is that, it depends on what question it is you are purporting to answer.
Last edited:
Scott, this is because of settings in Windows control panel and can be easily set properly and verified as well (loopback test). Hardware - sound - card used - playback - set to proper sample rate and bit resolution.
BTW, too much talking here.
No, you are making an unjustified leap in logic here - you simply don't know if the test itself has impaired the perceptual abilities of the listener & you have no internal controls to verify this
Matt
I heard from an E.B.U.transmittion Teodor Currentzis / MusicAeterna performing Beethoven’s 5th (Proms July 28th).
The intensity and dynamics of his style fitted perfectly the C minor key character of the symphony.
If you have access to BBC iPlayer, have a listen to it.
https://www.theguardian.com/music/2...na-teodor-currentzis-review-prom-18-beethoven
George
Ribbon mics are not used as much on vox as they used to be. But they can still be magic on the voice. KD Lang usually uses a Neumann U87 (hers) on her voice. But when recording the Case, Lang, Viers album a RCA 44BX was used on her vocals for the song Why Do We Fight. If you want to hear a stunning example of a ribbon on a vocal I recommend it, you can also listen to KD's normal U87 on the rest of her vocals on the album.
Ribbon mics have several characteristics that distinguish them from condenser mics. Notably the major resonance is extremely low, usually in the 20hz range or less, while condensers have their primary resonance in the top of the audio range, say 8-15khz. Lots of variables as to where including the capsule diameter.
Generally you can crank up a ribbon mic on the top end and it still sounds pleasant, while it is more of crap shoot on condensers. Inexpensive ones can have a horrid top when cranked, better ones can take it.
side notes on what we hear from the trenches
Frank Sinatra did use ribbon mics, early on, with Tommy Dorsey that was either a 44BX or a 77DX, later in his second career he started using the new condenser mics.
Cheers
Alan
Ribbon mics have several characteristics that distinguish them from condenser mics. Notably the major resonance is extremely low, usually in the 20hz range or less, while condensers have their primary resonance in the top of the audio range, say 8-15khz. Lots of variables as to where including the capsule diameter.
Generally you can crank up a ribbon mic on the top end and it still sounds pleasant, while it is more of crap shoot on condensers. Inexpensive ones can have a horrid top when cranked, better ones can take it.
side notes on what we hear from the trenches
Frank Sinatra did use ribbon mics, early on, with Tommy Dorsey that was either a 44BX or a 77DX, later in his second career he started using the new condenser mics.
Cheers
Alan
Last edited:
Thanks, I followed the links, unfortunately it looks like it's no longer available.
Both SY and Kevinkr have sent me "24 bit" files that were not. Audacity for instance in general does not in spite of settings. You need to examine the actual data 16bit LSB's in a 32 bit container is what you often get.
Read what fun people have... https://forum.audacityteam.org/view...sid=3ed45690c6ed73297024ee34b7ec30e6&start=40
48 kHz, WASAPI: This recording is simply perfect. The right channel's sine wave amplitude is below the 16 bit quantization threshold but even though it is bit-precisely recorded.
48 kHz, MME: 16 bit recording only with (obviously) some dithering noise added. I couldn't find how to stop that stupid dithering noise, particularly not in Preferences -> Quality.
96 kHz, WASAPI: Full failure, nothing to add.
96 kHz, Imported WAV-File: This import, recorded with a different program, is shown here just in order to prove that the interface works correctly.
96 kHz, MME: Works, 96 kHz with 16 bit only. Also the same stupid dithering.
192 kHz, MME: Just for completeness that the interface works here, too. But the source's sample width is in this case 16 bit only.
96 kHz, MME, Low-Pass Filtered: Interesting observation: Dithering must have taken place before truncation to 16 bit!
192 kHz, MME, Test Signal 0 to 90 kHz: This test signal was recorded in order to prove that Adacity in MME mode able to work correctly at 192 kHz.
Last edited:
No, just saying it is worth checking what foobar is actually sending out. On my machine only ARTA with paid for (not free) ASIO records and plays at real 24 bit. The free PortAudio library that many use does not do 24 bit, we went through this on another thread. When you ask people who don't fool around with this stuff normally to just use foobar you need to be careful because even sophisticated users have problems (see below),
https://hydrogenaud.io/index.php/topic,99707.0.html
NOTE - IIRC even ARTA does not do 24bit until you pay for it.
Last edited:
I asked the author of ARTA about what the differences are between the paid and freeware versions. He replied to the effect that the only difference is that some files may not be saved/written in the freeware version. In the audio device setup page in ARTA, the 'wav' bit-depth setting displays 16-bits and is grayed out in the freeware version. So far as I can tell, the sound card may be still accessed at 24-bits, such as for display of an FFT spectrum, but file I/O may be limited to 16-bits.
If someone wants 24-bit audio without paying, Reaper can do it. It can also use freeware bit-meter plugins to verify bit-depth. However even with Reaper one should use ASIO and verify the ASIO sound device is not set as the Windows default sound device.
Last edited:
I do check it. Foobar has always checked as perfect (of course in case you do not have some silly setting in preferences - output and DSP).
Audacity is a problem, to me it does not record in 96/24 but only in 96/16 even if it is set to 96/24. Same file from the same card recorded with SpectraLab is OK, 96/24.
http://pmacura.cz/tube_noise.zip
Audacity and CoolEdit Pro 2.1 are good tools to check the files. CE is for free in evaluation version, it just cannot save the files, but makes all analysis. On Y axis there is a division in sample numbers which is valid for 16-bit resolution. Level "1" means 1LSB 16bit. For a 24bit file, one must see sample values not only at integer 1,2,3 ...... etc., but also between them.
Sorry to interrupt a serious discourse of something I have no idea about, but let me again introduce what we were thinking about when most here were much younger, and many of us had responsible positions as engineers. In this example, Richard Marsh was working at LLL, and I was working at HK, and writing an IEEE paper (out a few months later) on cap distortion. Those were the days when we made progress in audio quality design!
Attachments
Since we all seem to want unlimited music storage and portability, computer memory technology is needed. So, we have to insert into the ADC and DAC complexity into what used to be an all analog audio system. This is where the new discoveries are being made.... how to improve them. All the above on 24 bit not really being 24 but often might be 16 etc is similar to what we did, John, in analog discoveries back in the day.
Like a C or an L isnt really the amount marked on the part at other frequencies besides 1KHz; Can't trust those numbers, either.
THx-RNMarsh
Like a C or an L isnt really the amount marked on the part at other frequencies besides 1KHz; Can't trust those numbers, either.
THx-RNMarsh
Last edited:
Nobody, for low cost. High cost, maybe. For one thing, most do not do high quality SRC, and SRC is needed whenever multiple clock domains are merged.
Actually, computers can be trusted, but it helps to have some experience and expertise with using them for audio. Say, don't you know someone like that?
yes, I do know someone like that....
But, I just want to listen to the music some time. You know, correctly done 24/96 Plug-N-Play? How can higher quality of audio and music hobby of ours ever get the attention of the next Gen of people with a situation like this?
Do we even know if all DBLT to compare 16b to 24b sound was not a 16 to 16 comparison?
Its a mess.
-RM
Last edited:
Yes I've been doing this for 20yr on Intel and Moto processors, I've had CE since I bought it in 2000.
I'm saying the average Joe is going to download Foobar and use the default settings and might not even know what ASIO is.
BTW bit depth on my Foobar is greyed out saying simply "appropriate one" will be chosen.
- Status
- Not open for further replies.