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is there any possibility of Ensoniq AudioPCI Wavesets (eapci2m.ecw / eapci4m.ecw / eapci8m.ecw) becoming supported in foo_midi/BASSMIDI? They were also used by some later SoundBlaster cards like Live! and Audigy, where the MIDI device named Creative SW Synth utilised them.
Failing that, are there any other apps I could use to play midis with these soundbanks?
Any word on this? (remaining faintly hopeful.)
I know there are others such as Coax/SPDIF, Optical/TOSLINK, and USB w/ each of them having their own benefits. I would think Toslink is much more forgiving if there is any problematic noise.Those are all digital so they wouldn't be used to connect the analog output from the DAC. If the integrated amp has digital inputs it has a built-in DAC (like your previous receiver).
You've mentioned frequency-response multiple times and does this relate to latency; for instance, with a player like fb2k and lowering the buffer-length lower than 1000ms or lowering latency for DJ'ing using mixer software like Mixxx?No. Frequency response is the variation in amplitude (loudness) as you move from low (bass) frequencies to high (treble) frequencies. That's a variation in the equipment... There are "natural" variations in music/program material. If you adjust the bass/treble controls or use an equalizer you are altering the frequency response. And equalizer an be used to correct for frequency response (to some extent) in your room/speakers or to improve a "bad recording". Or, you can use it to "enhance" the sound by boosting the bass, etc. to your taste.
Ideally you want "flat" frequency response across the audio range (20Hz - 20kHz) to accurately reproduce the recording. This is generally no problem for modern electronics. You can fairly-easily built an amplifier that goes from DC (0 Hz) to the MHz range. But, many amps intentionally filter-out DC because in the audio signal DC it can cause problems. Radio frequencies can sometimes cause problems too so usually the frequency response is somewhat limited to the audio range.
There is an upper limit with digital audio to half the sample rate. i.e. CD audio (44.1kHz sample rate) can't go over 22,050Hz and there is normally a low-pass "smoothing filter" so you don't get quite to that mathematical limit.
Speakers are a different story... Most speakers have a loss in the deep bass range and ALL speakers have up & down variations across the range. And room reflections have an influence on frequency response, especially the bass.
...The reason I mentioned frequency response is because it's one of the things you an actually hear and identify. If you say, "This DAC has more clarity" I won't know what you're talking about and I might be skeptical. If you say, "This DAC has weak bass" I might believe you and I'd know exactly what you're talking about.
Latency is delay. If you're just listening there's no issue with a several milliseconds of latency. I've got recordings that were made more than 50 years ago so a couple extra milliseconds don't matter. It can be an issue if you are watching a video and the audio gets delayed.
The main issue with latency is when you are recording yourself and monitoring in headphones. There is always latency through the computer and if there is too much latency it's hard to "perform". To eliminate that problem, some audio interfaces have zero-latency hardware-monitoring where the monitor signal doesn't go through the computer.
This now got me thinking - do DAC's help with volume and distortion?Again the distortion in modern electronics is usually measurable but inaudible, unless you overdrive a 100W amplifier, trying to get 150W out of it, etc. And again, speaker distortion is far worse than electronic distortion (under normal conditions). There is also a digital limit of 0dBFS* and you can clip (distort) digitally. The volume control on most software players can't go over "100%", so clipping isn't a problem as long as the digital recording itself isn't clipped. But, you can sometimes get clipping if you boost with a software equalizer. If there is digital clipping that would happen before the digital audio is sent to the DAC.
As far as "volume", there really is no standard for the analog-side. If there is a spec it should say something like 1V @ 0dB. Line level is rather loosely defined and it often depends on the loudness of the recording and a volume control somewhere in the signal path. But usually enough signal from the source and enough amplification in the amp or preamp so everything works together. And, you might have to adjust the volume when you switch between your DAC and your TV, etc. dBV and dBu are to electrical signal levels and there is no standard calibration standard between the analog-electrical levels and the digital levels.
* 0dBFS is the "digital maximum" so digital dB levels are usually negative. With integer formats it's the highest you can "count" with a given number of bits. Everything is automatically scaled before it's sent to the DAC so an 8-bit file is just as loud as a 24-bit file, and you can send a 24-bit file to a 16-bit DAC without clipping.
dB SPL is the acoustic loudness in the air. The 0dB SPL reference is approximately the quietest sound that can be heard so SPL levels are positive. There is a direct correlation between digital levels and acoustic sound level (a 3dB digital reduction results in a 3dB loudness reduction) but there is generally no calibration.
MOD edit: removed unnecessary quoted text
The HTTP 403 error started appearing again when attempting to play (music) videos on official accounts. Video on non-official accounts work fine.
I noticed that the 'SIDCC' parameter within the cookies info is updated with every request to YouTube, and the expiry date is set to the current date. This happens both in-browser and within foobar2000. The cookies.txt file is updated when calling videos via foo_youtube, so I assume this is expected behavior?
I used to be able to 'fix' this issue by restarting foobar2000. That doesn't work anymore. Renaming foo_youtube's cache directory to enable foo_youtube to recreate it and setting the user agent to the one my browser (Chrome, latest version) is using didn't work either.
[UPDATE] It just started working again, out of the blue. I did update the user agent once again (Chrome wasn't fully up-to-date after all).
Thank you, renaming the cache folder and then adding my browser's user agent fixed it.
I have a different question - how do I make the plugin completely ignore the uploader's channel name? Currently, it adds the uploader as "Artist" in the metadata.
The plugin also has an included "metadata pattern" which does a different thing - it parses the video title and tries to semantically guess the artist and add it to the metadata. But I'd rather it leave the video title as it is, and just ignore the uploader.
get a amplifier with network support ( dnla, airplay, spotify) and play happy from your chair , no more necessary to use dac or cables...
Convenient only if the audio setup is without constant change, sure. But, I dislike wireless because of it's ultimate frustrating procedures and protocols, then theres the wait time. Wires is forever reliable and not a problem for me and my experience with it compared to anything wireless.
Typically, the DAC has a pair of RCA outputs and the integrated amp usually has several pairs of RCA inputs. The inputs on the amp might be labeled "CD", "tape", "aux", etc., but they are all similar line-inputs. If there is a phono input (rare on modern equipment) it goes through a phono preamp.
This makes more sense now, thank you Doug. So, RCA's seem to be the common among most audiowireconnector-types. I know there are others such as Coax/SPDIF, Optical/TOSLINK, and USB w/ each of them having their own benefits. I would think Toslink is much more forgiving if there is any problematic noise.
Just to backtrack here this is how I plugged my Focusrite-scarlettsolo1stgen (audiointerface) to my Pioneer-vsx1022 (avr) from audiointerface's output to the avr's input. I think the audio-interface was damaged or defective in some way, maybe that explains the problematic static-hiss noise.
"Audiophiles" tend to use lots of meaningless words, so be skeptical when you read words that seem to mean something or words that put an image in your head but may mean different things to different people. What matters is noise, frequency response and distortion.* See Audiophoolery.
If you want "better sound", equalization (frequency response adjustment/alteration) is free in software, or better speakers always make a difference.
I believe this as I get more distracted by their "connoisseur'esque" descriptions than to benefit from hearing those rambles. You've mentioned frequency-response multiple times and does this relate to latency; for instance, with a player like fb2k and lowering the buffer-length lower than 1000ms or lowering latency for DJ'ing using mixer software like Mixxx? I hope this means no more delays as I get that with my PC's motherboard soundcard and I would absolutely love to eliminate audio delays/responsetimes.
This now got me thinking - do DAC's help with volume and distortion? And does Frequency-response relate to Audio-latency?
Though gnudb.org is an alternative source for the freedb database, it currently lacks features that will allow CTDB to retrieve and store the metadata. So for now you cannot submit new CDs for use by CTDB for freedb metadata.
You'll need create an account with MusicBrainz to submit new data.
You can insert the CD while CUERipper is running. Click the MusicBrainz icon (bottom right)
Once on the MusicBrainz page, click the "Submit this CD using the simple method" link and enter all the data.
Or use one of the other submission methods provided by the site, e.g.
Hey guys, apologies for the delays with updates. I hurt my neck almost 2 months ago and have been in a lot of pain and on a lot of painkillers. I can't sit at a desk or work very comfortably from a laptop so I haven't been able to get much done on this. I'd love to say a beta will be coming out before the new year, but at the moment I don't want to make any promises.Sorry to hear that, neck injuries are a huge PITA. Hope that it heals up nicely in the end.
I started out with the Xing MP3 encoder and was unhappy with the quality of resulting files.
That's when I stumbled upon r3mix and HA.
Here's wishing HA another 20 glorious years!