Quantcast
Channel: Hydrogenaudio Posts
Viewing all 11785 articles
Browse latest View live

Different Checksums

$
0
0
Sure, but who's to say there would have to be some indication of a problem? I would expect it to happen without any obvious sign of trouble other than the AR numbers. I also wouldn't assume there was any problem with the disc itself.

It could also simply be that there's a different pressing created with errant PCM data (EFM and CIRC are perfectly OK) which has no offset relative to the other version.

Finally, without comparing the two versions and identifying interpolatation, time shifts, bit-flips, dropped half samples and what ever other possible type of error, how is anyone to know which version is correct? Majority rule? I think that's for suckers (pointing to EAC's old bug when over reading is disabled).

Plain-text passwords in password reset

$
0
0
QUOTE (PostOfficeBuddy @ Aug 30 2015, 00:10) <{POST_SNAPBACK}>
This is a disservice to all users and fixing it is a worthy cause.


Yeah well SSL certificates still cost money and I didn't notice any offer to buy the certs in your post? Feel free to take your entitlement elsewhere.

Kudos to spoon for putting that money on the table. HA is on HTTPS now.

Different Checksums

$
0
0
QUOTE (pdq @ Sep 8 2015, 10:57) <{POST_SNAPBACK}>
QUOTE (korndawg @ Sep 8 2015, 13:43) <{POST_SNAPBACK}>
2) Will directly editing FLAC files within Audacity maintain audio quality? Should I disable dithering as well? Typically I only fire up Audacity in order to fade in/out live tracks.

You are not directly editing the FLAC file. Audacity first extracts the PCM data and operates on that. The final audio quality is a function of what you have Audacity do to the data.

If most of the track is not changed then I would disable dithering to avoid noise being added unnecessarily.

Yeah and without performing any operations, Audacity will also change the entire contents of a file if you load it and then save it when using the default settings.

Again, this really needs its own topic (not that it hasn't also been discussed on multiple occasions).

Get S/PDIF to playback properly at 37.8kHz

$
0
0
QUOTE (AlphaWaves @ Sep 2 2015, 16:35) <{POST_SNAPBACK}>
...

What I need is something to "resample" from 37K to 44K. I tried a few software like virtual cables but no-one was successful.

...


If you think it's samplerate issue in question then SRC should be done before sending it out from PS2.

If you can hear the audio sent from PS2 to Aureon through S/PDIF then the samplerate is OK but is it properly synchronized (if you look the wav data, there are few hundred samples long breaks after every few 1000 samples). Something disturbs the stream badly. Is there an option to set the sync source (OS options/PS2 options/Aureon control panel)? Have you tried to take the audio in using some software which has the option for S/PDIF clock source selection?

Also, common audio streaming issues can be hounted by DPC Latency tool.

What is for you the best album ever ?

$
0
0
Miles Davis-The Cellar Door Sessions

specific requirements - looking for compact model recommendation

$
0
0
QUOTE (zxc @ Sep 8 2015, 06:21) <{POST_SNAPBACK}>
QUOTE (audiophool @ Aug 28 2015, 09:01) <{POST_SNAPBACK}>
NTFS is a proprietary format. I'm guessing it may be prohibitvely expensive to provide NTFS support in an off-the-shelf micro system.

For example, the solution of NTFS-3G plus FUSE which is relatively common on non-Microsoft desktop systems is relatively CPU intense. So my guess is that whatever hardware is in such a micro system would have trouble handling NTFS.


all the smart tvs Ive ever used had no problems handling ntfs so my guess is that the same is possible with audio systems


Some Android devices will do NTFS, but not most. Anything old enough to be microsoft based will probably support it as well. But for newer products thats going to be tough. Maybe you can find an Android device that advertises NTFS (or test some in a store to see if the vendor licensed the NTFS format from Microsoft), but you'll probably have to look at a mini Windows PC of some kind if you must have NTFS.

Plain-text passwords in password reset

$
0
0
QUOTE (yourlord @ Sep 8 2015, 22:45) <{POST_SNAPBACK}>
Thanks for ponying up for the certificate Spoon. BTW, your spoon avatar image causes the site to be marked as insecure on threads you post in because it's an image hosted on a non-TLS site.


You'll see this in every thread where user content (typically img links) is posted too, because people tend to use http links. There's no way to fix this without forbidding it AFAIK, which I doubt HA wants to do.

IMHO Chrome's warning for this (striking through the https) is a bit obnoxious especially as they expressly allow passive content like images through in the first place (which is why you can see spoon's avatar). Firefox's warning sign+"attackers may alter the look of the site" message is a bit more on point.

HDCD "per track" decoding issue

$
0
0
Notice the whole lotta "seems" to follow, seemingly calling for comments, corrections, insults and maybe even (*gasp*) testing. I have posted fragments of this in other threads, after playing around with hdcd.exe / CUETools' HDCD features / foo_hdcd : https://www.hydrogenaud.io/forums/index.php...mp;#entry906496 and https://www.hydrogenaud.io/forums/index.php...9427&st=397

So I have advocated keeping a HDCD rip undecoded lossless for two reasons: (I) decoding is irreversible, and (II) one can decode on-the-fly upon playback. What I have found, indicates another argument against decoding (from tracks!), and one argument against on-the-fly decoding: the latter is that my decoding solution (foo_hdcd) is not bit-exact (see the latter of the above two links).


Now here is the hdcd.exe-decoding-by-tracks issue: hdcd.exe seems not to decode the very beginning of the input; only after it has read enough samples to identify HDCD.
That is: if fed individual tracks, it would switch off HDCD decoding briefly at the beginning of each.

Assuming that the goal of HDCD decoding would be to mimic actual HDCD playback through a DAC that is fed an SPDIF signal without track boundary information, the "good" thing to do is to start decoding as soon as HDCD is identified in track 1, and then go on for as long as a HDCD-aware DAC would do.
(Presumably for the entire CD if done properly?) So we have a case for image-based conversion.

But some CDs only have a subset of tracks as HDCD. And it seems to me that hdcd.exe only scans the first 750 frames. I conjecture it has no way of switching off HDCD. That is an issue against converting a full image. One should then use hdcd.exe -i to identify each track, and then decode-as-one each "contiguous set of HDCD tracks"?


It seems that CUETools - which uses the .dll version of the hdcd.exe utility, but works inherently on full CD rips (and can enforce reading past 750) - is immune against the beginning-of-track issue.




Samsung's UHQ upscaler

$
0
0
Is Samsung's UHQ upscaling algorithm in their newest mobile phones the stuff that Harman's Clari-Fi branch offers, or have they developed their own thing? Does anyone know?

(btw, I know this stuff is bullshit, I'm interested in the business relationship behind the scenes...)

aoTuV Patches, Vorbis 1.3.5 and Lancer

$
0
0
It is in ICC15.

But if john33 is using ICC14 you have compile in VS2010 to get XP support.

Disadvantages to linear phase low-pass filters?

$
0
0
Yes, you will get some imaging above Fs/2 with such filters.

PCM1794a sharp filter has similar specs but 130dB stopband attenuation.
WM8742 has several different filters to choose from, some reach stopband at 0.5 Fs.

With your numbers a full-scale 20 kHz sine (an artificial signal that you won't find in music) would appear as image at 24.1 kHz at <-75 dBFS.

HTTPS is now supported on Hydrogen Audio

$
0
0
QUOTE (marc2003 @ Sep 9 2015, 14:21) <{POST_SNAPBACK}>
how about automatic re-direction of http > https? that doesn't appear to be implemented at the moment but is usually standard when a site supports https.


You'll get redirected when you just enter the domain name. Not if you follow an HTTP link to inside the forums. Maybe the latter can be done if it's verified nothing important is broken.

HT systems vs two-channel for music listening

$
0
0
You Born Again zealots are something else.

QUOTE (mzil @ Sep 8 2015, 23:35) <{POST_SNAPBACK}>
Wrong! All it will do is add 3 dB, end of story. As long as the weaker amp is kept within its operational range for the entire program material

What if it isn't and is driven to clipping peaks? Still Wrong! ?

QUOTE (mzil @ Sep 8 2015, 23:35) <{POST_SNAPBACK}>
including the loud passages and peaks, then it will be audibly indistinguishable from the stronger amp.

What if it clips on those peaks, still indistinguishable to your strawman?

QUOTE (mzil @ Sep 8 2015, 23:35) <{POST_SNAPBACK}>
The onset of distortion and full on clipping is rather quick with solid state amps, and although it is true you may only notice it on the louder peaks at first, as long as you don't dial it up to that point at which it distorts on anything, then you are good to go.

What if you do dial it to that point? Still good to go?

QUOTE (mzil @ Sep 8 2015, 23:35) <{POST_SNAPBACK}>
I wonder what his explanation is as to why science has never discovered a machine that can measure this elusive "extra adrenaline/clean sound" he speaks of, which for some odd reason disappears in DBTs

What DBTs does audible clipping disappear? Cite them please.

QUOTE (mzil @ Sep 8 2015, 23:35) <{POST_SNAPBACK}>
If one is distorting on the loud peaks, simply back down a dB or two on the volume knob and you're back in the perfectly clean, safe area.

Or add a more powerful amp and listen as loudly and cleanly as you desire. Oh but wait, you use to sell those scams, so now must crusade against them in penance. Ooops, sorry, "AVR Hater", forgot. rolleyes.gif

Data on 320CBR vs LOSSLESS

$
0
0
QUOTE (pdq @ Sep 9 2015, 14:03) <{POST_SNAPBACK}>
QUOTE (uchihaitachi @ Sep 9 2015, 08:58) <{POST_SNAPBACK}>
Problem is I use itunes and it doesn't have native support for FLAC. And ALAC is annoying, I guess I will stick to 320cbr

Out of curiosity, what is annoying about ALAC?


Just not as universal as FLAC?

I'm confused

$
0
0
This is in reply to your previous post.

In post #3, Case already addressed the other question. Yes it is possible and it's a bit of a stretch to call upsampling the lower res track improper, unless you use a junky SRC. I'd probably not use Gold Wave just to be on the safe side.

Now that you found your lowest transparent setting...

$
0
0
well since i use a PSP i use nero aac at 170k since I've gotten better results than other encoders I've used.

Disadvantages to linear phase low-pass filters?

HDCD "per track" decoding issue

$
0
0
It takes many samples to detect an HDCD signal (or the loss thereof), and during that time, the decoder must use whichever HDCD settings were previously set.

When you play a whole CD, the start of each track will be played using whatever HDCD settings were enabled at the end of the previous track, until the decoder is able to detect the new settings or loses the HDCD signal and turns all HDCD features off. This is why the differences you see are always at the beginning of a track.

If your goal is to imitate an HDCD-capable DAC, you should perform full-image HDCD decoding for all of your CDs, regardless of HDCD content. (HDCD requires extra headroom, so it reduces the volume of everything else by about 6dB.)

QUOTE (Porcus @ Sep 9 2015, 01:19) <{POST_SNAPBACK}>
And it seems to me that hdcd.exe only scans the first 750 frames.

Only when you use the -i flag. If you don't use the -i flag, it will check the entire file and correctly report HDCD even if no HDCD is present for the first 750 frames.

Data on 320CBR vs LOSSLESS

$
0
0
QUOTE (uchihaitachi @ Sep 9 2015, 05:58) <{POST_SNAPBACK}>
QUOTE (apastuszak @ Sep 9 2015, 05:11) <{POST_SNAPBACK}>
QUOTE (clintb @ Sep 8 2015, 21:51) <{POST_SNAPBACK}>
QUOTE
Thanks for the response.

I am just trying to standardise bitrates for my music library. It's all mixed ranging from 192 to 320 to FLAC. Is 320kbps CBR a good choice?

If you're looking to standardize, keep everything you can in FLAC and transcode to a target device. Anything you have in lossy format, leave it that way.


Gotta agree here. Keep it in FLAC to give you the freedom to transcode to anything in the future.


Problem is I use itunes and it doesn't have native support for FLAC. And ALAC is annoying, I guess I will stick to 320cbr

thanks!

Are you using Windows or Mac? If using Windows, stick with FLAC for an archive, transcode to something lossy for usage in iTunes. Done. Heck, same thing on Mac.

I know iTunes is a fairly nice player, and manages moving music to iDevices smoothly as well; keep it that way. For the love of all that is holy in music, keep your sources lossless. Seriously. I started on this quest back when Dibrom created this site years ago after a falling out at r3mix and made mistakes along the way with ripping/archiving/etc. Above all else, I learned to keep it lossless.

Backing up a bit, are you embarking on a ripping project?

Successful ABX of 88.2 kHz vs 44.1 kHz

$
0
0
Regarding the sample rate dependent sound quality issue, a famous example is Creative 10kx soundcards (Live/Audigy) having very different measured results in 44k vs 48k.

Another not so famous example is Asus Xonar D2 (the first Xonar). The measurements in ixbt and even the official Asus pdf report showed obviously more inferior performance in 44k, even though it is unlikely to be audible.
http://ixbtlabs.com/articles3/multimedia/asus-d2.html

See page 14-15
http://audio.rightmark.org/downloads/Xonar...stGuide_V12.pdf

The result of Xonar D2 in 44k is strange because unlike Creative's cards, Xonar D2 is supposed to support 44.1/88.2/176.4k clocks natively, as shown in the specs
https://www.asus.com/Sound-Cards/Xonar_D2PM/specifications/
Viewing all 11785 articles
Browse latest View live