It's a quick question. I'll add why I'm asking, but it's not necessary to read it, because I'm really a newbie and may have many misconceptions for a single thread.
I've noticed all AC3 tracks from my DVDs show cut at 20 kHz on Spek, and I thought they were cut that way to better take advantage of the limited bitrate, storing data only on the audible range.
Then, I found that the only DVD-Audio I have sounds better on the MLP part than on the AC3 one, and I'm trying to investigate why is that, because I'm a skeptic on Hi-Res audio. I converted an MLP track to ̶A̶A̶C̶ ALAC, only because Spek won't open MLP, and to AC3 (448 and 640) to compare if it sounds better than the original AC3 448. And I found what I'm showing in the image. AC3 limited to 20 kHz. selecting both 48 and 44,1 kHz samplig rates, and both files having the exact same size. Then, I converted to DTS and the spectrum goes up to the expected 24 kHz. I used Xmedia Recode
Why have a 48 or a 44,1 Khz sampling frequency, if they end up limited at the same 40 kHz?
+ Reply to Thread
Results 1 to 17 of 17
Last edited by MLP; 27th Apr 2020 at 10:52.
Spek is perhaps limited to 22 kHz, at least such maximum I saw for files which go over it. MLP is lossless compression for PCM, while AC3 is lossy. Unless you are bat , you won't hear deference between lossless audio cut at or well before 20 kHz, like FM at approx. 16 kHz. Higher sampling frequencies have different purpose than to carry a sound - it's to reduce intermodulation distortion and to preserve more dynamics.
Lossy audio formats like AC3, MP3, or AAC will limit the amount of bitrate spent on the higher frequencies to be efficient. The population that can hear above 20khz is fairly limited, so why waste bandwidth up there is what the encoder is thinking. If you up the bitrate the encoder might fill up the spectrum or it might not. Depends on the encoder.
Spek seems to be working fine by the way.
Spek itself ain't limited, it will show 96khz if loaded. Ac3 and what not will cutoff at certain values depending on bitrate and channels.if all else fails read the manual
Thank you all for your replies. I'm answering to all of you tomorrow, but first...
Mmm, guys? I converted an MLP to wav with XMedia, and that wav to ac3 with XMedia and Aften. Both resulting .ac3s weight the same 21.5 MB. Is XMedia removing the ultrasonic frequencies in order to take full advantage of those 640 kbps on the audible range? I think that's cool, but confusing, being that it hasn't an option to select to do that or encode the real 48,000 o 44,100. Maybe it's just cutting off and wasting space. But Aften cuts off below 20,000 if I select 512 or 448 kbps.
The highest frequency which can be represented in theory without distortion (aliasing) is 1/2 of the sampling frequency, means 22.05kHz for 44.1kHz sampling rate or 24.0kHz for 48kHz sampling rate, for example. For a practical implementation it is somewhat less due to the roll-off of the anti-aliasing lowpass filter.
AFAIK the official Dolby Encoder cuts all frequencies above 20.3 kHz. ATSC specifies 21.75kHz for 44.1kHz sampling rate, and 23.67 kHz for 48kHz sampling rate. Aften AC3 encoder goes as high as 23.67 kHz for 48kHz sampling rate.
Last edited by Sharc; 25th Apr 2020 at 18:03. Reason: Added ATSC
xmediarecode uses ffmpeg. You can set the -cutoff for the ffmpeg ac3 encoder. eg. -cutoff 24000
But I don't know if xmediarecode allows you to enter custom commandline arguments
This screenshot a stereo AC3 384kb/s encode , but it works for 192kb/s for this example too
[Attachment 52890 - Click to enlarge]
[Attachment 52891 - Click to enlarge]
Last edited by poisondeathray; 25th Apr 2020 at 19:08.
Why 44.1KHz? Why not 48KHz?
None of that means a given codec necessarily uses a 20kHz cut-off frequency by default. The high stuff is expensive to encode in respect to bitrate (and the idea of a lossy encoder is to compress the audio), especially as most people's hearing doesn't extend to anywhere near 20kHz. The default cut-off frequency for the LAME MP3 encoder is around 18kHz, but I think most encoders adjust the cut-off frequency according to the chosen bitrate (or quality for variable bitrate encoding).
The spectrum for your Apple Lossless encode seems a bit odd, because if it was encoded from a 48k lossy source, it shows very high frequencies that shouldn't exist. I don't know why that'd be happening, aside from maybe some crappy upsampling. I don't really understand the difference for DTS either.
Original EAC3 (5.1ch)
Re-encoded as AC3 with foobar2000/ffmpeg at 640kbps
Re-encoded as DTS with foobar2000/ffmpeg at 1536kbps
Re-encoded as lossy AAC with foobar2000/QAAC at v91
It appears the default true variable bitrate quality uses a slightly lower cut-off frequency.
Apple Lossless at 96k/24bit, upsampled with foobar2000's dBpoweramp/SSRC resampler DSP
Most HD audio is a complete waste of time. It's simply non-HD audio such as older analogue recordings remastered and upsampled to a bitdepth and frequency greater than required, and labelled as HD. True HD audio needs to be at least 24 bit and 96k throughout the entire recording and mastering chain, and of course it needs to be the same for the entire playback chain. This guy agrees and he should know what he's talking about. He claims true HD audio is easier on the ears and the brain. I've never had the opportunity to listen to any, but I suspect it'd require state of the art equipment and speakers to hear a difference, even if it appears to be something of a subconscious difference.
Is that why my cat happily snoozes with the TV volume cranked up? What sounds hi-fi to me might sound like muffled background noise to him.
Aften, as recommended here, Spek goes up to the 24 kHz limit. Now I learnt these AC3s are intentionally cut at 20 kHz.
You are kind explaining things, but I'm not so kind talking about Hi-Res audio. I'm not a bat, nor are the people who created the music, and even they don't know how these ultra high frequencies their instruments may produce are supposed to sound. Neil Young is at the same time a promoter of Hi-Res, and an environmental activist. Don't wasting Internet bandwidth and storage media creating files that weight 100 times what is necessary aggravates global warming? Like, having to use 100 memory cards for what can be hold into one alone?
I'm in favor of digital compression. I think more efficient compressing, creating smaller files that sound transparent is the way to go.
Do higher sampling rates really make a difference apart from ultrasound? Or are you referring to bit rate?
Last edited by MLP; 27th Apr 2020 at 11:12.
̶O̶o̶p̶s̶,̶ ̶I̶ ̶f̶a̶i̶l̶e̶d̶ ̶a̶t̶ ̶q̶u̶o̶t̶i̶n̶g̶.̶ Fixed.
Last edited by MLP; 27th Apr 2020 at 11:18.
My screenshots were all taken after converting the audio using the same EAC3 source.
ALAC is a lossless format (Apple Lossless), whereas AAC is lossy. I encoded an ALAC version because your original screenshot showed ALAC (I upsampled the 48k source to 96k while converting), and I also encoded a lossy AAC version for comparison. I have no idea why XMedia gave you ALAC if you chose AAC.
I guess the difference in the higher frequencies in your screenshots just relates to the cut-off frequency of the low pass filter and how steep it is for the different codecs.
Out of curiosity... you said the MLP version sounds better than the AC3 version from the disc? After you convert the MLP version to AC3 or DTS yourself, does it still sound better than the AC3 version from the disc? I'm curious if it sounds better because MLP sounds noticeably better than standard AC3, or if it sounds better because it was mastered differently.
As a side note.... I just tested a 96k sample rate with the various AAC encoders. They all encoded the audio as 96k except for QAAC. I even added --rate 96000 to the command line and the encoded audio was still 48k. Anyone know why that is?
Adding 2 cents:
1) Converting an ac3 source (e.g. from a DVD) via "lossless" to any other fancy audio format using encoder xy is pretty much pointless. The ac3 "original" has already removed all frequencies above approx. 20 kHz, and what is lost is lost and cannot be recovered by reencoding. It's the same mistake people make when upscaling low resolution video sources to HD or UHD resolution. One doesn't get more picture details, just bloated files.
High audio sampling rates make sense for studios producing "masters" for further processing and multimedia distribution. These "masters" should catch any frequencies which might be present, as well as as the full dynamic range and environment impact etc. of the original "natural" analogue sound. The target is "not to loose anything" to the extent what the available technology allows.
Bitrate is in linear proportion to the sampling rate. It is also driven by the dynamic range (nearby jet engine vs. falling leave, acceptable quantization error/noise floor ....) and by the number of channels (mono->stereo->5.1->7.1 etc.), and of course a matter of the lossy compression algorithm.
I think the converted MLP to AC3 448 kbps sounds the same as the original MLP, better than the original AC3, especially on the voice.
I didn't want to add confusion, but one of my mini plug to RCA cables broke and the electronic stores near me are closed. I'm using S/PDIF, converting everything on the fly wirh DTS Interactive and Dolby Digital Live. My receiver didn't even have HDMI. So, I cannot make a comparison that would be taken seriously. But, don't these spectrograms look different? I don't know how to read them, maybe they show the converted file just sounds louder. Or even more compressed? If I had to guess, I'd say the converted file looks more compressed. I'm looking forward to converting the MLP with a 20 kHz cut-off. I'm examining my DVDs with DTS an AC3 tracks that sound pretty different to me, and I'm viewing that their spectrograms also look different,