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Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
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91
CUETools / Re: How would you interpret my accurate rip results?
Last post by korth -
Quote from: anteater545 on
Also, would CueTools "Fix" my track #13 even though it matched some people's? Or how would that work?
There's only one recovery record for that CD (CTDBID DD7BC2BE, current confidence 139)
CUETools should repair 33 samples
  6  | (139/161) differs in 31 samples @04:08:58-04:08:60
 13  | (139/161) differs in 2 samples @03:03:44

Quote from: anteater545 on
Track 13's length is 3:04:38 and the 2 different samples are at 3:03:44 so it isn't just an offset kind of problem I assume. I just want to rule out any problem with my hardware or setup.
You could post a complete extraction log

Quote from: anteater545 on
If I use the repair tool, I find when checking with accuraterip, some of my songs are +12, and some are -12. Is it using 2 separate recovery records? Before the repair, they're all +0 for the ones that wrre accurare.
You're not posting logs so we can see what you're referring to. Repair would only change the samples I referenced above.
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92
3rd Party Plugins - (fb2k) / Re: ReplayGain DSP - Alternative ReplayGain implementation by Case
Last post by Mekoz -
I use the playlist tabs and drag my music to them, they can be a mixture of old and new tunes. I use replaygain to get a general loudness level.
 
I use the DSP drop-down list to change profiles.

Looking at the console im not seeing replaygain mentioned all that often.

I have noticed one instance when playing a song from another tab/playlist that replaygain seems to ignore the song and it plays louder than it should?
93
CUETools / Re: How would you interpret my accurate rip results?
Last post by anteater545 -
Quote from: Kartoffelbrei on
what results do you get for accuraterip? This is CTDB. I usually just call it good when one of the two checks out. ^^

Id guess that it would fix ur track but with those results i wouldnt know if the fix would actually be more correct. If one checks out then i would suspect some ocd stuff that is not relevant to musical integrity. Then again i dont know what you are after.

It got moved to the CTDB forums.

Accuraterip reports the same results.

I just think it's strange that I got same read and test CRCs, followed by matching 7 other rips, and now I get read errors on track 13. I thought matching even 1 was considered proof of a good rip, but apparently not.

I'm after a bit-perfect cd rip.

If I use the repair tool, I find when checking with accuraterip, some of my songs are +12, and some are -12. Is it using 2 separate recovery records? Before the repair, they're all +0 for the ones that wrre accurare.
94
General Audio / Re: Live stream format with proper metadata & embededd album covers
Last post by ktf -
Quote from: nu774 on
Can you add FLAC_free() or something in favor of Windows ?
In Windows, apps and DLLs can be linked to different C runtime libraries. In such cases, If a DLL allocates memory with malloc() and an  app attempts to free that memory by free(), bad things can happen (usually SEGV).
Ah, that is a bummer. I wasn't aware this could be necessary. Do you know of another project with such a wrapper, so I can take a look whether there are any additional caveats?

Also, if you have suggestions for improvements, please let me know. This addition has seen little review, and I'm unsure whether the current set of functions is enough to be implemented easily, or whether more functions are necessary.
95
Listening Tests / Re: A killer sample for Opus 256 VBR
Last post by Case -
I have been meaning to ask if you who hear difference could share what you use to listen to the files. The lowpassed file shared here looks perfect, the difference between it and the original only contains very quiet high frequency tones. It's completely inaudible to me even using headphones at loud volume. The signals are nowhere loud enough to cause clipping, even intersample peaks are about 1.3 dB below digital fullscale.
My logical mind says the files should sound the same, unless the difference signal is also somehow audible for you.

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96
Listening Tests / Re: A killer sample for Opus 256 VBR
Last post by magicgoose -
Yes that's what I would think, too. Aliasing or nonlinear distortion somewhere in the playback chain after the decoding.
A similar, more extreme case can happen when playing audio sourced from a SACD (DSD) which hasn't been lowpassed, there is so much ultrasound noise.

> Sox 20000hz, lowpass width: 5%, Phase response: linear

Yeah that should be good enough by far.
97
CUETools / Re: How would you interpret my accurate rip results?
Last post by Kartoffelbrei -
what results do you get for accuraterip? This is CTDB. I usually just call it good when one of the two checks out. ^^

Id guess that it would fix ur track but with those results i wouldnt know if the fix would actually be more correct. If one checks out then i would suspect some ocd stuff that is not relevant to musical integrity. Then again i dont know what you are after.
98
General Audio / Re: Live stream format with proper metadata & embededd album covers
Last post by nu774 -
Quote from: ktf on
Actually, this functionality has been added with FLAC 1.5.0. Along with multithreading, this was the other major new feature.
I didn't know this either, and I found rich set of new APIs in FLAC/stream_decoder.h for chained ogg stream.

Just one thing...

Can you add FLAC_free() or something in favor of Windows ?
In Windows, apps and DLLs can be linked to different C runtime libraries. In such cases, If a DLL allocates memory with malloc() and an  app attempts to free that memory by free(), bad things can happen (usually SEGV).

FLAC__stream_decoder_get_link_lengths() requires exactly this, which is problematic.
To avoid this issue, DLL can simply export a wrapper of free() function of the C runtime library to which DLL is linked.
100
General Audio / Re: Opus vs FDK-AAC in 2025
Last post by a.ok.in -
Quote from: btc on
Quote from: Kamedo2 on
Yeah, Opus codec lacks Spectral Band Replication(SBR), the patent which would help in going very stingy on bitrate, so Opus understandably and predictably struggles at 32kbps or lower. On the upside, Opus is simple and computationally cheaper than HE-AAC.
Opus lacks SBR as it is was patented. Correction, patents related to SBR expired. While I am not a lawyer, all SBR-related patents mentioned online appear to have expired. As a general rule, after 22 years from a format's approval, defending the existence of related patents becomes difficult. Anyways, 20+ years later all interest in low rates, SBR and such things are all long gone.
Instead of SBR, Opus uses band folding. Jean-Marc was quite modest and didn't state that band folding is superior. Unlike SBR, which resamples audio to 22kHz from the original 44.1kHz, limiting its use to lower bitrates, band folding maintains higher quality and remains effective even at bitrates of 128 kbps and above. 

Opus bandwidth extension is not as simple as it appears. As @jensed said band folding in conjunction with spectral energy envelop outperform SBR.

Quote from: Kamedo2 on
... Opus understandably and predictably struggles at 32kbps ...
Opus low performance at low rates is due to frequency leakage (low delay), not due to lack of / or weak bandwidth extension.

There's a thing called AAC-ELD, it's basically a low-delay HE-AAC, built over AAC-LD (which already starts to sound too muffled below ~64kbps, so adding SBR was needed to keep a good bandwidth). According to this graph its algorithmic delay is ~20 ms like Opus. There's also AAC-ELDv2, which adds some low-delay form of Parametric Stereo. I'm not using ELDv2 as I don't have any way to encode it... it seems that it could be a little better than Opus for practically most low-bitrate cases, though having a slightly higher delay. (AAC-ELD only with SBR is sometimes better, many times worse or equal to Opus at the 32-48 kbps edge; above 48kbps Opus is definitely better. I suppose that below 32kbps AAC-ELDv2 would be the best option... afaik xHE-AAC isn't designed for low-delay)
 :>
I attach a 7zip file with six lossless audio samples, and AAC-ELD (m4a) / Opus audio files at 32kbps. Just in case there are decoded (flac) versions of the m4a files ("fdk-aac packet decoder" plugin is needed for playback of AAC-ELD on foobar2000).