Quality and BitRate of SAT Recordings posted on tribal mixes
its not worth the arguing anyways, i know about dish network feeds for sirius, because i did a little study on it.. it proves factual that sirius feeds dishnetwork a 192 constant bit rate feed... now as far as XM and direct tv.. i have no clue.. all i said was i wanted proof of the quality for XM and DirectTV before i started wasting space on my hd's... i am limited as is, and if the quality REALLY is 320, then im all for it... but if you are NOT recording from DIRECTtv, then i HIGHLY DOUBT you are getting a feed that strong, as most feeds on XM are ONLY 60kbps to the receiver...
oh well enough about this issue....
oh well enough about this issue....
Of course, one thing is all about sample rates, bit rates and so on ...
... but the other thing is how to record for example a live stream properly. I mean how to record to avoid too much loudness + limiting. It's a shame that many radio stations are broadcasting streams with very high compression/limiting which means low dynamic range
= poor, squashed, distorted sound
loudness race
Of course it makes no sense to record a bad quality source in highest bitrate but it makes no sense too to record a stream not properly. Maybe we can have a general discussion not about "ripping" but about recording live streams/sets properly ?
... but the other thing is how to record for example a live stream properly. I mean how to record to avoid too much loudness + limiting. It's a shame that many radio stations are broadcasting streams with very high compression/limiting which means low dynamic range
= poor, squashed, distorted soundloudness race
Of course it makes no sense to record a bad quality source in highest bitrate but it makes no sense too to record a stream not properly. Maybe we can have a general discussion not about "ripping" but about recording live streams/sets properly ?
I appreciate the discussion an of course I will participate in it.So, what is the problem? Now we all agree that behind the "waste of bandwidth and time" the real problem is the “Quality and BitRate of SAT Recordings”.
I would define it more restricted as a “Quality of the XM Sat Radio Cannel 80 Broadcasts”. And that is because I can not talk about “Quality and BitRate of SAT Recordings” in general but I can talk about “Quality of the XM Sat Radio Cannel 80 Broadcasts” in particular.
There would be no problem if there were data about the XM Sat Radio Broadcasts themselves but as it is clear these are proprietary data and as such we have not them.
So, how to evaluate the Quality of the XM Sat Radio Cannel 80 Broadcast? As long as we have the result of the broadcast – data representing a sound - the natural approach is to try to evaluate the quality of the sound and to make some conclusion about the topic of interest.
The evaluation of a sound quality is not a simple task. The approaches so far were based on some expert evaluations. Personally, I do not think the opinions shown were really expert evaluations and I do not agree entirely.
What is my approach?
I have real data – files containing digital representation of a sound. I can use formal mathematical methods to process the data and to receive some understandable and interpretable results and then to compare them with analogues results obtained from data representing sound which recognized experts have already evaluated, let me say, at least as good CD quality sound. And if I can say the comparison shows the formal results, obtained by formal strict mathematical transformations are alike I could conclude that the sound quality was alike also.
To do that, I think, I can use as an expert opinion CDs already on the market released by recognized as experts labels and artists. I could work with many but what I use is the CD Album: GU002 OAKENFOLD - NEW YORK (Label: GLOBAL UNDERGROUND).
I believe the album should be at least a “Good” quality sound. I made a precise CD rip with the wonderful application named Exact Audio Copy. So, these are my data, and I think, they can be rated as at least a “Good” sound quality.
The data to compare are, of course, the data recorded from a XM Sat radio Channel 80 broadcast.
Now, what we actually need when evaluating the sound quality? We need to find the proper compression so as to preserve the sound quality. With a loss compression like an mp3 compression the lost of quality is inevitable. And again, how to evaluate the losses?
One possible approach is to use the results of a Frequency Analysis – spectra. It is relatively easy to implement, the method is accessible – every sound editor has such a tool, so the results are easy verifiable.
To clarify this before going in details let me mention the idea of “bandwidth” which is widely used. The signals and particularly the sound signals (analog and digital), considered as mathematical functions of the time, can be transformed into other mathematical functions – spectra- with a parameter “omega” or frequency. Not strictly defined, the bandwidth of a signal is the range of possible changes of the frequency parameter of its spectrum function. The bandwidth is defined for the technical devices also, as the bandwidth of the signal they can process without changing it. For example: the speech signal has bandwidth approximately 3-4 kHz. That is why the GSM standard defines a bandwidth such that music can not be transferred without essential losses over the GSM cannels.
Now, if the spectrum of a signal has been preserved when compressing it, this means the signal has not been changed and so the sound quality has been preserved also. And opposite, if the spectrum has been changed, let’s say by reducing the bandwidth this means the signal has been changed and so the sound quality has changed also but it has been made worse.
Not going further in more details, I would like to continue with some general discussion.
The CD quality sound has not a strict definition but the CD recording parameters are strictly defined by Sony/Philips standardizing documents (“Red Book” as long as I remember.) So, I would say the CD quality sound is a sound which quality is restricted only by the CD recording parameters. And the restrictions are: Two channel, 44,100 digital samples per second, and 16 bits representing a digital sample. Again not going into details, I would like to say: 16 bits impose noise which can be reduced using 24 bits to represent the digital samples, and 44,100 samples per second restricts the recorded signal to having no more than 22,050 Hz bandwidth. In other words a CD quality sound should be represented by a signal having a bandwidth of 22 kHz or close to it, let’s say no less than 21 kHz. One more thing to mention about the bandwidth importance. The professional digital audio devices use 48 kHz, 96 kHz, 192 kHz as a sampling rate. Having in mind a human can hear sound wave with a frequency let’s say up to 22 kHz why it s necessary to sample a signal at a rate higher than 44.1 kHz when this rate guaranties the analog signal, that actually we hear, can be fully recovered from its digital representation? The answer to me is that the researches have found the high frequency components in the signal spectrum are important and they have to be presented more precisely. That is why I stress on the signal bandwidth.
Finally, I would like to say: my investigations on XM Sat Channel 80 Broadcasts have shown me the broadcasts have the CD quality the way I understand it. But I need to say that they are not the perfect CD quality sound and the broadcast itself shows some drawbacks also. I’ll show them later but I am not going to explain them. Maybe they are results of some processing methods used to reduce the amount of the information transferred to and from satellite but they can be used for effectively compressed the recordings at lower bitrates.
Now I am ready to prove that the XM Sat Channel 80 Broadcasts audio quality is a CD audio quality and compressing them in mp3 format at 320 kbps is not a "waste of bandwidth and time”. But I’ll do this later on in another posting. I need to make some drawing over a couple of graphics representing the mathematical functions – spectra, so as to help understanding the proof, and to upload them.
Skype:spas.velev
well u have real data, the oakenfold cd, and it is ready to analyse or for encoding to mp3, flac etc. thats okay.
but i think u cant compare it with the XM data, because it comes highly compressed and encrypted and than proprietary decoded.
that is not the same basis, isnt it?
i guess this is the difficult part here. as long as there is no way to find out what the real specifications of the stream are, one have to live with the PCM output.
and by not having the raw stream, every transcoding station generates additional noise...
and ur absolutely right: quality of XM sat recordings is the main point!
i agree that just a listening test with a couple of interested peeps could bring a near idea of a solution.
here just another note from one of the developers of mp3
and it is a law of business to minimize the total costs of ownership where possible.
who is thinking XM has no 16khz frequency cutt off?
but i think u cant compare it with the XM data, because it comes highly compressed and encrypted and than proprietary decoded.
that is not the same basis, isnt it?
i guess this is the difficult part here. as long as there is no way to find out what the real specifications of the stream are, one have to live with the PCM output.
and by not having the raw stream, every transcoding station generates additional noise...
and ur absolutely right: quality of XM sat recordings is the main point!
i agree that just a listening test with a couple of interested peeps could bring a near idea of a solution.
here just another note from one of the developers of mp3
Karlheinz Brandenburg wrote:
Read the following text about bandwidth by Karlheinz Brandenburg
from MP3 and AAC explained :
The bandwidth myth
Reports about encoder testing often include the mention of the bandwidth of the compressed audio signal. In a lot of cases this is due to misunderstandings about human hearing on one hand and encoding strategies on the other hand.
Hearing at high frequencies
It is certainly true that a large number of (especially young) subjects are perfectly able to hear single sounds at frequencies up to and sometimes well above 20 kHz. However, contrary to popular belief, the author is not aware of any scientific experiment which showed beyond doubt that there is any listener (trained or not) able to detect the difference between a (complex) musical signal with content up to 20 kHz and the same signal, but bandlimited to around 16 kHz. To make it clear, there are some hints to the fact that there are listeners with such capabilities, but the full scientific proof has not yet been given. As a corollary to this (for a lot of people unexpected) theorem, it is a good encoding strategy to limit the frequency response of an MP3 or AAC encoder to 16 kHz (or below if necessary). This is possible because of the brick-wall characteristic of the filters in the en-coder/decoder filterbank. The generalization of this ob-servation to other types of audio equipment (in particular analog) is not correct: Usually the frequency response of the system is changed well below the cutoff point. Since any deviation from the ideal straight line in frequency re-sponse is very audible, normal audio equipment has to support much higher frequencies in order to have the required perfectly flat frequency response up to 16 kHz.
Encoding strategies
While loss of bandwidth below the frequency given by the limits of human hearing is a coding artifact, it is not necessarily the case that an encoder producing higher bandwidth compressed audio sounds better. There is a basic tradeoff where to spent the bits available for encoding. If they are used to improve frequency response, they are no longer available to produce a clean sound at lower frequencies. To leave this tradeoff to the encoder algo-rithm often produces a bad sounding audio signal with the high frequency cutoff point varying from block to block. According to the current state of the art, it is best to introduce a fixed bandwidth limitation if the encoding is done at a bit-rate where no consistent clean reproduction of the full bandwidth signal is possible. Technically, both MP3 and AAC can reproduce signal content up to the limit given by the actual sampling frequency. If there are en-coders with a fixed limited frequency response (at a given bit-rate) compared to another encoder with much larger bandwidth (at the same bit-rate), experience tells that in most cases the encoder with the lower bandwidth pro-duces better sounding compressed audio. However, there is a limit to this statement: At low bit-rates (64 kbit/s for stereo and lower) the question of the best tradeoff in terms of bandwidth versus cleanness is a hotly contested question of taste. We have found that even trained listeners sometimes completely disagree about the bandwidth a given encoder should be run at.
Read the following text about bandwidth by Karlheinz Brandenburg
from MP3 and AAC explained :
The bandwidth myth
Reports about encoder testing often include the mention of the bandwidth of the compressed audio signal. In a lot of cases this is due to misunderstandings about human hearing on one hand and encoding strategies on the other hand.
Hearing at high frequencies
It is certainly true that a large number of (especially young) subjects are perfectly able to hear single sounds at frequencies up to and sometimes well above 20 kHz. However, contrary to popular belief, the author is not aware of any scientific experiment which showed beyond doubt that there is any listener (trained or not) able to detect the difference between a (complex) musical signal with content up to 20 kHz and the same signal, but bandlimited to around 16 kHz. To make it clear, there are some hints to the fact that there are listeners with such capabilities, but the full scientific proof has not yet been given. As a corollary to this (for a lot of people unexpected) theorem, it is a good encoding strategy to limit the frequency response of an MP3 or AAC encoder to 16 kHz (or below if necessary). This is possible because of the brick-wall characteristic of the filters in the en-coder/decoder filterbank. The generalization of this ob-servation to other types of audio equipment (in particular analog) is not correct: Usually the frequency response of the system is changed well below the cutoff point. Since any deviation from the ideal straight line in frequency re-sponse is very audible, normal audio equipment has to support much higher frequencies in order to have the required perfectly flat frequency response up to 16 kHz.
Encoding strategies
While loss of bandwidth below the frequency given by the limits of human hearing is a coding artifact, it is not necessarily the case that an encoder producing higher bandwidth compressed audio sounds better. There is a basic tradeoff where to spent the bits available for encoding. If they are used to improve frequency response, they are no longer available to produce a clean sound at lower frequencies. To leave this tradeoff to the encoder algo-rithm often produces a bad sounding audio signal with the high frequency cutoff point varying from block to block. According to the current state of the art, it is best to introduce a fixed bandwidth limitation if the encoding is done at a bit-rate where no consistent clean reproduction of the full bandwidth signal is possible. Technically, both MP3 and AAC can reproduce signal content up to the limit given by the actual sampling frequency. If there are en-coders with a fixed limited frequency response (at a given bit-rate) compared to another encoder with much larger bandwidth (at the same bit-rate), experience tells that in most cases the encoder with the lower bandwidth pro-duces better sounding compressed audio. However, there is a limit to this statement: At low bit-rates (64 kbit/s for stereo and lower) the question of the best tradeoff in terms of bandwidth versus cleanness is a hotly contested question of taste. We have found that even trained listeners sometimes completely disagree about the bandwidth a given encoder should be run at.
and it is a law of business to minimize the total costs of ownership where possible.
who is thinking XM has no 16khz frequency cutt off?
Hi,Now I am showing spectra obtained from a CD rip and from XM recording. Both files containing the sound data were in .wav (Microsoft Wave) format which means the data are uncompressed representing Pulse Code Modulated (PCM) at 1,411.2 kbps audio signal (two channel, 44.1 kHz sampling rate, 16 bits per sample). To generate the spectra I have used the Sony Sound Forge 8.0’s Spectrum Analysis tool.
Let me first clarify. The tool uses Fast Fourier Transform (FFT) which in turn calculates the spectrum based on limited number of samples. I use the maximum number which is 65536. This means the spectrum is calculated over 1.286 sec period of time. Because of this short time period the spectrum can vary over the time depending on the sound generator (artist). That is why the better way to study the sound quality is by using a Sonogram. For my purpose, I thing, using the spectrum is enough. To make the spectrum more representative I select a sound of duration of about 3 min to be used for FFT calculations. My goal is to have some sort of “averaged” spectrum over a longer time period. The real goal should be the extreme parameters: the max frequency component, the max dynamic range but I am not going to write a new Frequency Analyzer to reach such a goal. So, I use something I consider an acceptable evaluation of the spectrum. It could be better, in sense of quality, but not worse. Then if the results of the procedures are acceptable for this spectrum they can not be worse for the whole sound record also.
I have looked at the the (01) MYSTICA. Bliss (Mystica Mix).wav spectrum (CD rip).
The first question I have asked myself was “What is the spectrum bandwidth”. The way it looks acceptable to define it was to accept that the quantization noise should be small compared to the minimum value presented in the spectrum. The “small” is relative idea, so after some attempts I have chosen a “small” to be 100 times less than the minimum spectrum value presented in the range of the bandwidth. This means, in terms of the software calculations which I am not going to explain, -104.4 dB or spectrum bandwidth of about 21.3 kHz. Without being very precise we could accept a spectrum band of 21 kHz at level -104 dB.
Now take a look at the graphics. They represents the spectra of two sound files: the CD rip - (01) MYSTICA. Bliss (Mystica Mix).mp3 and a XM record - Sound 4.wav. Can you tell which one is a CD rip and which one is XM Sat Radio record? And if you can answer positively I would like to explain the differences that helped you to answer the question. Do not think I do not see differences. I have marked them but I am not going to discuss them.
Next thing to do is to have a closer look at the high frequency spectrum range – the range over 10 kHz and to compare how different mp3 compressors work and what is the difference between 192 kbps and 320 kbps compressions.
Skype:spas.velev
I saw the previous post and I have to agree. Yes, we can not do anything but to use the PCM output of the satellite receiver. Not going in details the receiver SNR is > 80 dB which is very, very good.And, I am saying The XM cut off frequency is 22.05 kHz not 16 kHz.
Skype:spas.velev
Now, you can see and make a visual qualitative comparison between uncompressed and compressed sound files spectra.
I am not going to compare the different spectra using quantitative methods although it is possible. The only thing I would like is for you to have some idea and qualitative evaluation.
While the mp3 compression at 320 kbps preserves the spectrum of the original data other bitrates obviously do not. Again, I am not evaluating the losses quantitative and I am not considering the topic in general.
The worse compressor is Sony’s Plug-In 2.0 which reduces the spectrum to the 16 kHz even on the assumption that the ratio Minimal Signal Level to the Quantization Error is 20 dB (or 100) which correspond to level -104 dB (marked in red). But if we want to impose more strict condition of this ratio being 46 dB (or 205) then the signal level should be above the level of -96 dB. This would lead to a result that both mp3 compressors at 192 kbps would reduce the signal spectrum to 16 kHz. I am not sainig anything about what does it mean. I am only pointing out the reducing in the spectrum bandwidth.
A couple of words about high frequency components in the sound signal spectra.
I can not discus this topic in essence because I do not know it. But what I know for sure is:
1) One of the effects of increasing the sampling rates in the digital audio is representing the high frequency components more truthfully.
2) The high frequency components in a complex sound signal actually form the rising and dropping edges of the sound wave. How important it is I do not know but they are not a sound to be hear at.
Skype:spas.velev
Now I am ready to prove that the XM Sat Radio Channel 80 broadcasts have an audio CD qualityWe can use any method you want.
Skype:spas.velev
SpasV wrote:
Now I am ready to prove that the XM Sat Radio Channel 80 broadcasts have an audio CD quality
We can use any method you want.
Now I am ready to prove that the XM Sat Radio Channel 80 broadcasts have an audio CD qualityWe can use any method you want.
CD quality by most COMMERCIAL companies is 192.. they dont count on us ppl that know every little twitch and tweak of the sound...
i said i was done with this topic, but, i still am waiting on actual PROOF that XM from even direct TV or straight from the SAT are higher than 192.. not based off your machine or anything like that..
and by the way.. why are your uploads only @ 192 now.. and not 320????
Hi,1) The process of proving has two sides. On one hand there is a proof on other there should be an understanding. I do not now what proof you can understand. So, tell me what proof do you need.
2) Jimmie Page and Steve Lawler - at UFSR on The Move (XM Sat) - 19-Feb-2007 @256 kbps.
I will continue to upload the sound from John Digweed, Carl Cox, the Pete Tong show, the Jimmie Page show and some others compressed as mp3 at least at 256 kbps.
And I will use 192 kbps compression for reson that has nothing in common with the higher quality sound.
Skype:spas.velev
looks nice spasv 
regarding the PCM out i would like to see a record from an ordinary SAT channel that is also
available maybe one with the common DVB-S MP2 audio stream.
unfortunately i off my pc actually. but im back next week and i will post some 'analysis' from my ASTRA satellite :)
btw: is there a noticeable difference in listening to the wav, 320, 192 version? i mean for you in personal. i would like to make a kind of listening test with some peeps, should be interesting.
regarding the PCM out i would like to see a record from an ordinary SAT channel that is also
available maybe one with the common DVB-S MP2 audio stream.
unfortunately i off my pc actually. but im back next week and i will post some 'analysis' from my ASTRA satellite :)
btw: is there a noticeable difference in listening to the wav, 320, 192 version? i mean for you in personal. i would like to make a kind of listening test with some peeps, should be interesting.
SoCoJosh wrote:
CD quality by most COMMERCIAL companies is 192.. they dont count on us ppl that know every little twitch and tweak of the sound...
well they advertise as digital and not cd-quality, this talks for itself!CD quality by most COMMERCIAL companies is 192.. they dont count on us ppl that know every little twitch and tweak of the sound...
SoCoJosh wrote:
i said i was done with this topic, but, i still am waiting on actual PROOF that XM from even direct TV or straight from the SAT are higher than 192.. not based off your machine or anything like that..
stay tuned, we're on it i said i was done with this topic, but, i still am waiting on actual PROOF that XM from even direct TV or straight from the SAT are higher than 192.. not based off your machine or anything like that..
SoCoJosh wrote:
and by the way.. why are your uploads only @ 192 now.. and not 320????
please tell us if u hear a difference!and by the way.. why are your uploads only @ 192 now.. and not 320????
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the xm sat stream is: 64kbit/s to 96kbit/s HEC aacPlus v1 and has no cd quality!
if one wants no know more, read on the mess starts right here:
hoi there
first: okay i take all the blame for starting a useless technobabel discussion.
but i HOPE im not the only one concerned about the posted stuff here!
and i dont want anyone to stop uploading stuff here, i just want bring some light in about how stuff works.
okay take a look here to this torrent and the comments:
https://www.tribalmixes.com/details.php?id=13145&tocomm=1
there is a SAT rip encoded lossless with FLAC. WTF!
i hope im not the only one seeing the picture that this is just a waste of hdd-space.
the statement from the uploader
not a "waste of bandwidth and time "
it leads me to suppose that there is not much technical background. :(
a second reply was:
I use Palk XRt12 tuner with direct digital outputs (PCM format) and its optical output, connected to the optical input of the sound card (SB Audigy).
I use recording software which records from Sound Blaster Audigy - S/PDIF-In (direct digital input) and generates an WAV (uncompressed) Audio type file with Sample rate: 44.1 kHz, Bits/Sample: 16 bits, Cannel: Stereo, and Bit rate 1,411.2 kbps (which actually is a PCM format).
The editing software, I use, has a Frequency Analysis Tool, based on FFT (Fast Fourier Transform), which shown me a Frequency bandwidth of the processing digital signal more than 20 kHz. After having compressed the .wav file to .flac and decompressed it back the Frequency Analysis Tool shown me the Frequency bandwidth again more than 20 kHz.
So, I really think the files are true lossless compressed with the quality of the broadcasted signal.
P.S. According to my FATool the Frequency bandwidth of a 192 kbps mp3 file is less than 15 kHz and the Frequency bandwidth of a 320 kbps mp3 file is less than 20 kHz (usually around 19 kHz).
and when there was an aac source, where came the >20khz from?!
read more here: https://jthz.com/mp3/CD-44100Hz.htm
i tried to bring some light in with:
@spasv
aehm...what does all the bold stuff mean
you are totally wrong. im sorry to say that.
all signals comming from: satellites, cables, fm radio and internetstreams are in somewhat commpressed codec format. that is a fact! do a research on wikipedia or howstuffworks!
considering that means: making a 'lossless' file from that is a waste of space, time and bandwidth!
if u really want to be a ripper u should get a digital satelllite card for ur pc and save the RAW stream comming from the sky.
that will be in most cases a 192kbps 48khz mp2 audio stream.
suprised?
this one u can transcode to an mp3.
i hope this helped a bit.
i admitt, at this time i had no clue that XM is ENCRYPTED!!! WTF!!!
so u have to use a proprietary decoder and only have a PCM out.
no RAW SAT source, what a pity...
then there was this reply:
“aehm...what does all the bold stuff mean?”
It means two things.
First of all, I assume the XM Satellite Radio’s statement they broadcast CD quality sound and the Polk Audio XRt12 Reference Tuner’s digital output in PCM format mean my raw input signal is a 1,411.2 kbps bit stream representing a two channel 44.1 kHz sampled 16 bits digitalized sound wave.
Second, by recording this stream in a .wav file as a two channel 44.1 kHz sampled 16 bits digitalized Pulse Code Modulated signal and compressing it to a .flag file I have fully preserved the quality of the original Carl Cox at Global show broadcasted on The Move cannel of XM Satellite Radio on 05-Jan-2007.
P.S.
First, I do not have any doubt about my assumption.
Second, thanks for your teaching me how to record a satellite broadcast. I would suggest that you first do it for yourself and if you can do it compare your solution with what I have already shown.
they advertise 'digital' and this means nothing!!!
i have a simple statement from the avsforum.com
Neither service is CD quality, online or on a radio. Also neither service advertises that they are CD quality, they advertise 'digital' quality which may be seen as misleading, but it is true. My cell phone is digital, but the SQ is worst than awful.
U can read more here: https://www.avsforum.com/avs-vb/showthread.php?t=751216&page=1
and lukily i found this today:
https://www.tribalmixes.com/details.php?id=14672
i question the bitrate.. and no i have not downloaded the set...but i know that sirius does not do higher than 192, and they have the most updated system of the 2.... so, as in espresso's part, im going to have to say there might be some wasted megz on this set.... GREAT upload tho.. dont get me wrong.. just dont see the need for 320kbps...
sorry man..
unfortunately there was just this reply:
Of course, it is your choice whether or not to download the files.
I would like to say a couple of things though.
1) It is impossible to discus a topic such as Quality of a Digital Music in a comment like this.
2) It is impossible to discus a topic such as Quality of XM and Sirius Satellite Radio Broadcast in a comment like this.
But it is possible for me to say that the Sony MP3 Plug-In 2.0 used as an mp3 encoder with Sony Sound Forge 8.0 can not compress an original high quality CD sound better than as at 192 kbps even at the highest possible mp3 bitrate of 320 kbps.
Why am I saying this? For two reasons.
1) Because when compressing at 320 kbps I do not use the Sony’s software.
2) I can not say anything about Sirius Sat Radio but as long as you mention “espresso“ I could say that as long as I am informed the uploader “espresso1967” uses for his compressions Sound Forge 7.0. This simply implies it was possible for his compressed files to be compromised by the Sony’s software.
Finally, you said “... but i know that Sirius does not do higher than 192”. I do not think you know that.
More over, I am sure you do not know that. And of course do not get me wrong.
this is again nothing, sorry no offence but thats it. XM and Sirius both have no CD quality as a fact!
XM is just a joke because they dont publish their bitrates! this is on a need-to-know-basis. just LOL!!!
so lets talk peeps!!! and im always interested on some new technical infos
btw: i do SAT Rip myself on european ASTRA sats via RAW digital stream capturing and no funny PCM transcoding ...