Quality and BitRate of SAT Recordings posted on tribal mixes, page 4

really this is the most complicating topic i have ever seen in TM i feel tht spasv and tomix are genuies i really dont understand anything from this topic but it has been a long one so long  

I’ll take the “artificial high frequency” for a while.
First, as Stefan Meltzer and Gerald Moser say, sorry I have to repeat that again, “In traditional audio coding, a significant amount of information is spent in coding the high frequencies, although the psychoacoustic importance of the last one or two octaves is relatively low. This triggered the basic idea behind SBR. Based on the cognition of a strong correlation between the high- and the low-frequency range of an audio signal (hereafter referred to as the “high band” and the “low band” respectively), a good approximation of the original input signal high band can be achieved by a transposition from the low band”
So, they have proven there is a strong correlation between the high- and the low-frequency range of an audio signal.  What actually is artificial high frequency”? This is the very nature of the sound.
Second, unfortunately I can not go in details at this time, but the MPEG-4 HE-AAC v2 (also known as “aacPlus v2”) is a MPEG-4 profile (HE-AAC v2). In other words, the world wide audio compression experts have already accepted the aac+ v2. I do not see any reason for me not to accept it also.
Third, there is an essential difference. My approach is to preserve the sound quality without reducing it by the file size. And I control the sound quality using formal methods to evaluate it. The arguments against this approach are: “There is not a sound quality in the XM Sat radio broadcast better than one corresponding to a sound compressed at 64 kbps, the implication is mp3 compression,  bit stream” So, no need to use 320 kbps mp3 compression. I have not seen any proof about that, only guessing and free assumptions. But the spectra I have shown prove the sound wave of the XM Sat radio broadcast is preserved much more better by an 320 kbps mp3 compression than by the commercial 192 kbps. And an argument such an “artificial high frequency” can not be an argument. It is a high scientific achievement . Do not underestimate it.

please remember xm uses a non standard aac codec!
they have their own proprietary one!
and do i have to repeat the Karlheinz Brandenburg qoute?!
didnt u realized that there is an all over ~15 - 16khz cutoff? it is also visible in ur graphs!
the rest that is visible beyond the 15khz is the artificial part read on.
now regarding the 'artificial high frequency' you have asked:
so the stuff above 15khz is artificial because it is approximated-genereated and has nothing to do with the original source!
and guess what is one of the reasons:again there is the Brandenburg one
in plain text: humans can hardly to not hear differences above ~15 -16khz

and no i dont want you to encode the xm stuff at 64 or 96kbit/s cbr mp3! hell no!
but take the time and do the Ojay method!

maybe u can see the difference but u cant tell apart while listening to it!

This is not an argument also. It could serve you when making your choice to download or not to download some file. So, this is your freedom – to have a choice. But is not an argument and there are a couple of reasons easy to mention:
First of all, a subjective evaluation is not so simple as it seems to be. A Google search would show a number of international recognized and approved test methodologies for listening tests. For example (https://en.wikipedia.org/wiki/MUSHRA) “MUSHRA stands for MUltiple Stimuli with Hidden Reference and Anchor and is a methodology for subjective evaluation. It is defined by ITU-R recommendation BS.1534-1.” (Here ITU stands for International Telecommunication Union.) And “A codec listening test is a scientific study designed to compare two or more lossy audio codecs”. So, do not underestimate the topic.
Second, there are at least three resons that can compromize the subjective evaluation:
• the personal hearing
• the quality of the sound system in use
• the backgrounnd noise level

WTF! no argument? lol
humans are no binary beings ... whatever ur graph is saying ...

these test are already done. remember? thats why the cutoff at ~16khz is all around.
and because hec aac+ v1 is that high compressed they have to do artificial, approximated, estimated, propietary additions to the high-band in their decoding devices that has nothing to do with the original source but is it transparent for the users. they cant tell the difference, they are human
and that was determined by highly sientific listening tests.

so ur encoding at 320 to preserver ur full artificial information and the rest of the world is using 'Lame -V 0 --vbr-new --lowpass 16' or the more common 192kbit/s CBR way
okay u have the right to do so, even if there is no need to...

but dont try to sell ur encoding as 'super high quality' as ive seen it here

i have the feeling you dont want to see that this xm shit is nothing special at all.
do u really believe it has better quality than a stream?!
no it has not! it is almost the same maybe equal and has a lot more artifacts!
the only advantage is it uses a much lower bandwidth and the profit for the companies can grow!!!
and plz consider that we are talking about EDM and not very high demanding classic music!

im not convinced by ur given data to go off my statement, that encoding the crappy xm signal at 320kbit/s cbr is a waste of time, bandwidth and hdd-space.
and by taking the raw facts:
a 64 - 96kbit/s hec aac+ v1 stream as source one can judge for his own...

please see this debate as a try to save ressources by reducing hdd-space-usage, dl-time, bandwidth-usage, pc-uptime and so on ...

@SpasV: Both Sirius (PAC) and XM (AAC+) with their low bitrates can't be enhanced in quality by encoding the streams in 320kbps MP3.

Did you ever see a river flow backwards, from lower elevations to higher ones? No? The same is true for audio streams - you can't get a higher quality by encoding a low bitrate 64kbps source at high bitrate 320kbps. In contrast, the 320kbps version is of even lower quality than the original broadcast because codecs such as MP3 are lossy...

@arnani: if you intend do record / rip from satellite radio it is good to know all this to avoid a few common mistakes....

@Ojay: The main conflict point in this discussion is the strange assumption about “a low bitrate 64kbps”. It is not clear where it comes from at all. Someone has said, someone has heard. This is not a serious discussion.
All the time I have used strict frequency analysis showing that the XM satellite receiver output signal spectrum can not be generated by a digital signal corresponding to a bit stream at bit rate less than 320kbps. And no word about these analysis’s.

Now I have a few interesting spectra to show. All of them are generated using a CD rip made by EAC from a Global Underground release.
First, these are the spectra of the uncompressed file (.wav) and the compressed at 192kbps (.mp3) file using Lame compressor.
As it can be seen the essential differences are over the high band (16kHz-22kHz) range of the spectra.


That is why I provided 6 more spectra displayed in the range 12.00kHz – 22.05kHz .
I am not going to comment them right now, maybe later on. What I would like to point out is:
• It is clearly seen an mp3 compression at 192kbps can not preserve the original signal spectrum. It can well preserve around 16 kHz band.
• It is clearly seen an compression at 320kbps can almost preserve the original signal spectrum.
• It is clearly seen the AAC (m4a) compression is better than an mp3 the compression.

One more thing to mention: It is seen again that if the signal has a dynamic range of 57 dB and spectrum bandwidth of 22.05 kHz it can not been transferred by a bit stream having a bit rate less than 320 kbps. Then, it follows that if the XM Sat receiver output signal has such characteristics the communication channel is equivalent to a bit stream at least at 320 kbps.

If you really want to preserve the full spectrum of frequencies up to 22.5 kHz with MP3 files, use the following lame command line:

'lame -V 0 --vbr-new -k'

It is the -k switch that does the trick. That is as good as 320kbps, you get the full frequency spectrum and the size is only a fraction of the 320kbps one.

You can do the same trick with 192kbps CBR MP3 files but it isn't recommended as you loose bitrate for the lower frequencies while you can't hear the highest frequencies....

@spasv
hec aac+ v1 is a low bitrate codec! only developed to serve acceptable quality at the lowest bitrate possible. i think u have missed that point. it is the main reason for developing such codecs!
regarding ur statement hec aac+ v1 surpasses mp3 by a factor of 3,33! this is what u are trying to tell here!
show me one source telling the same...

i am confused!
what do you assume? you are capturing a PCM signal and try to reconstruct the data of the stream!

fuck xm! the bitrates arent published by them! this tells all!

and please dont try telling us ur graphs show the true data about the xm stream because they cant! they show a captured PCM source!  

connect ur sat-dish to ur pc, capture the encoded xm sat-stream (no need for decoding here)
stop the time past, divide the captured data by the past seconds and you have your bitrate!

and the phrase 'low bitrate 64kbit/s' comes from the developers of the aac+
because their aac+ delivers cd like quality at 64kbit/s!
noone has heard, nobody has told, it comes from the very developers.

First:
Hi, do not take me wrong. Our discussion reminds me a game. You give an exercise and I try to show that “to think you know something is different from really knowing it”
There are other spectra. The problem I am giving is to find out the mapping between the spectra and the next commands:

lame -cbr -b320 -k mystica.wav CBRmystica.mp3
lame -v --vbr-new -k mystica.wav VBRmystica.mp3
lame --preset insane mystica.wav insane.mp3
lame --preset insane -k mystica.wav insane-k.mp3

Here is a hint: you can not have a better mp3 compression than that defined by command line parameters:
-k             keep ALL frequencies (disables all filters)
-b320      specify minimum allowed bitrate  
(-B     specify maximum allowed bitrate, default 320 kbps)

Second:
Now I see the misunderstanding. “you are capturing a PCM signal and try to reconstruct the data of the stream”
The PCM signal I record is an already reconstructed, from the data carried by the stream through the satellite channel., digital signal.
What I do is analyzing the digital signal I record in a wav file container and showing “it is impossible to have such spectra” from data corresponding to a stream at rate less than 320 kbps.
As to XM ... i would say the same.

yes i see the missunderstanding too:
xm broadcasts 64 - 96kbit/s (u can check with the method given above) to ur reciever,
the decoder decodes the hec aac+ stream to a PCM waveform,
and is adding artificial data to the highband of the PCM waveform,
u record the PCM waveform with the original stream data plus the added artificial highband data
by the additional data the spectrum is artificially broadened
and this finally brings us to the point that ur analysis shows a full spectrum covering PCM waveform even better than cd
that in return has to be encoded with the highest bitrate possible bitrate in mp3 to cover the whole spectrum.

to me this is a simple naive miscalculation!
scientists have prooven that the highband above 16khz can be ignored.
business depends on the science!
aac+ maybe has a real cuttoff at 12 - 14khz before broadcasting and an artificial reconstruction
at the endpoint that generates no bandwidthusage. thats the way business goes.