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Topic: About XM radio spectrum (Read 6219 times) previous topic - next topic
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About XM radio spectrum

 
Is there something new I could say?
Not very much and maybe not important for most of you.
But …
First of all let me say a couple of introductory words about a signal spectrum for those who are not familiar with it.
The sound is a perception caused by a sound wave - a wave of pressure that spreads through the air. (The sound wave spread not only through the air.)
So, there are two points to look at the sound:
1.   One is the point of the human perception. I am not going it to this topic at all.
2.   The second is the point of the physics, mathematics, and engineering. This is the point I am interested in.
The sound wave can generate a signal which can be treated as a function. In digital sound we are dealing with discrete functions. Discrete functions are sequences of samples taken from the continuous functions (analog signals) at equally spaced moments of time. A continuous function can be perfectly reconstructed by its discrete derivative under some conditions that are usually met.
The sound is very complex and for its studding a Spectrum Analysis is used. The Spectrum Analysis of discrete functions is based on FFT (Fast Fourier Transform is a method for fast and effective calculating the Fourier Transform of discrete functions) named in honor of the French mathematician Fourier who first proposed the idea for a complex function to be consider as a sum of simpler functions maybe around 200 years ago. The Fourier Transform is unique and reversible which means that if you have a discrete function its Fourier transform is unique and if you have a Fourier Transform of a function you can calculate the function back also. There are many such transform that are in use. One of them is MDCT (Modified Discrete Cosine Transform) which is used to compress the discrete functions and the sound in particular also.

Here is a spectrum of a sound. The sound is the track #11 from Tiesto's album ISOS 5. I use this track because its source is reliable. In other words if a sound has a spectrum which is similar I would consider it good.




The spectrum is shown in logarithmic and linear scale.

Now about XM radio broadcast.
What is known about it?
XM use AAC + v 2 as an encoding standard.
What does it mean? It means AAC (Advance Audio Coding), accepted in MPEG-4 also with m4a audio format, and SBR (Sub Band Replication).
The  m4a sound format, and accordingly AAC, is already known. What is SBR? SBR is patented method for sound reconstruction using information about its spectrum.

Knowing these and something more about DSP (Digital Signal Processing) let us look at the sound spectrum of an XM broadcast.




Looking at the first spectrum I could say it is similar to the one above - of a CD track sound.
Looking at the second spectrum - it is not so similar. The essential differences are in the middle frequency range: 5,000 Hz - 15,000 Hz and in the high frequency range: 15,000 Hz - 22,050 Hz.

Let us look at them closer.




What is first seen is the slope in the middle frequency range. It is very important because if we have it almost flat, as it is in the ISOS spectrum, the high frequency range would be better also.
Let me say it. What can be done about making the sound better is just that - equalizing the spectrum. It is not a trivial task though. So, I left this topic aside for a while.
Second, there are drops in the spectrum at 5.5 kHz, at 11.0 kHz, and at 15.0 kHz. They are not natural and my understanding about them is:
The drop at 15.0 kHz is maybe due to SBR. In other words - the frequency range 20 Hz - 15 kHz of the signal is encoded using traditional approach (AAC encoding) while the high frequency ranges 15.0 kHz - 22.05 kHz using the SBR approach. Or the signal is first split in two frequency ranges: 20 Hz - 15 kHz and 15 kHz - 22.05 kHz.
Then:…
Then the range  20 Hz - 15 kHz is split in three ranges: 20 Hz - 5.5 kHz, 5.5 kHz - 11 kHz, and 11 kHz - 15 kHz. The reason for this my understanding is the two drops in the spectrum at 5.5 kHz and at 11 kHz. These drops are result of an imperfect signal reconstruction when using so called QMF (Quadrature Mirror Filter - a method for transfer data through a communication cannel). The method allows reducing the bit rate needed to transfer the data exploiting the decreasing spectrum amplitude vs. the frequency. To get more advantage of the method the spectrum is shaped with a slope about 20 dB/decade in the range 5.5 kHz - 15.0 kHz (this slope is missing in a CD track's spectrum).
In other words this part of the spectrum is shaped to have the mention slope then split in three parts. Every part then quantized using actually an ACC encoding and because of that information is lost the signal spectrum has these drops when reconstructed again at the receiving side.

There was a question (Tom Mix asked it): what is the encoding bit rate? It is impossible for me to determine it at that time.
Because there are three sub bands in the range 20 Hz - 15 kHz to encode it is possible almost every thing and I do not have means to evaluate that. I have the reconstructed signal only and I cannot find reliable differences to distinguish when encoding and decoding a signal with spectrum bandwidth of 15 kHz @65 kbps or @400 kbps using AAC encoder. So, I think it is possible for the 20 Hz - 5.5 kHz band to be encoded @32 - 64 kbps, for the 5.5 kHz - 11 kHz band to be encoded @96 kbps - 128 kbps, and for the 11 kHz - 15 kHz band to be encoded @128 kbps - 192 kbps or something more suitable.

But what I can do is to equalize the signal spectrum.
Without going into details I do equalizing in 6 - 9 steps using the Sony Sound Forge.
Because the procedure is time consuming I am going to write a filter (a computer program) to process equalizing automatically.
The result of equalizing is shown next.






The first and the third are the ISOS 5 track's spectra and the second and the fourth are the equalized XM radio's spectra.
As it can be seen the signal spectrum of the XM radio broadcast now is very close to the spectrum of a CD track. It is almost very good. Maybe I'll try to shape it better but I am not thinking of correcting the two drops at all.
Because of this spectrum I have for the XM tunes I encode my uploads as VBR m4a @320 kbps. Actually I use quality factor of 0.8 which results in bit rate around 320 kbps. The maximum quality factor of 1.0 encodes a CD track @400 kbps.

You can download some of my equqlized files here:

ASOT Live from Amsterdam
http://www.megaupload.com/***

ASOT Year 2007 Mix
http://www.megaupload.com/***

Moderation: removed links to copyrighted material.

About XM radio spectrum

Reply #1
Fascinating! It never occurred to me that an equalizer could restore a highly compressed (both data and loudness) audio file to close to original quality!

About XM radio spectrum

Reply #2
You don't need to lecture us on what a power spectrum is. Moreover, you've provided no details on the technical configuration of your spectral analysis. I see no description of what musical material was used for the XM analysis, although I can only hope it's the same song that you analyzed from CD. The graphs would be much easier to visualize if both sources were shown on the same plots.

Analyzing one song is not conclusive enough for the conclusions you are making. For all you know, XM is using a different mix. They are certainly using far more processing than mere equalization, such as multiband compression, that may affect different songs in different ways - and moreover that processing is impossible to overcome through additional processing, much less additional eq.

About XM radio spectrum

Reply #3
Quote
What does it mean? It means AAC (Advance Audio Coding), accepted in MPEG-4 also with m4a audio format, and SBR (Sub Band Replication).

SBR means Spectral Band Replication.
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About XM radio spectrum

Reply #4
David Ranada, who was technical editor of Stereo Review in the 1990s, did an extensive comparison between an original CD, an XM broadcast, and a Sirius broadcast, and published several pages of spectral analysis and listening tests.

His conclusion was that both XM and Sirius were using dynamic range compression to keep the signal levels as constant as possible, and then encoding a variation of AAC in the 32kbps-48kpbs range. Ranada cited several obvious artifacts, which anyone can hear with the right programming.

Ranada's conclusions were that XM sounded slightly better than Sirius, and that Sirius had both worse sound quality and more reception problems (the latter due to Sirius' satellite positions). My own experiences exactly mirror Ranada's.

I would add that XM and Sirius tend to change bitrates depending on the audience numbers listening to each programming channel. For example, the traffic information channels on the higher-numbered stations are extremely poor, probably using data compression in the 16kbps range (barely tolerable for voice). The classical and jazz stations seem to have the best overall sound, but it depends on the specific channel.

Sirius claims to offer 128kbps streaming feeds via their website (at an added fee), but I haven't tested to see if the sound quality is really better or not. They have added a feature where, if you don't click a button on the website after a certain amount of time (several hours), the streaming feed shuts down -- which is a drag if you're trying to record any of their programming unattended.

To answer the O.P.'s question: because the type of processing that XM and Sirius is very complex and uses a lot of psychoacoustic cues for encoding, simple EQ will not fix the sound problems their low bitrates cause. Maybe if Sirius and XM finally merge, they can whittle down the duplicated channels and then increase the bitrate on all their programming, providing better-quality sound. As is, their stuff sounds pretty bad. A good FM station sounds far better than XM and Sirius in every way except possibly for S/N ratio. The stuff I've heard has heaps of distortion compared to, say, a decent 128k MP3.

BTW, both companies use uncompressed 44.1 WAV files for all their in-house material, stored on sets of 20TB servers, and do the encoding as the signal hits their uplink. They claim they have the ability to change encoding methods on the fly, and that all existing receivers will automatically decode higher (or lower) bitrates.

About XM radio spectrum

Reply #5
Maybe if Sirius and XM finally merge, they can whittle down the duplicated channels and then increase the bitrate on all their programming, providing better-quality sound.

That is if the companies ever cared about quality at all.

Quote
What can be done about making the sound better is just that - equalizing the spectrum.

Your spectrum shows fake (repeated) HF data due to SBR. No eq can bring the actual data back. You'll have the same "volume" of signal there but of completely different "structure".

About XM radio spectrum

Reply #6
 My idea was to identify the characteristics of the XM radio broadcast and to show what is correctable.
Thanks for replies.
I need to add an replication.




The spectra are shown on the same plots. Blue - the XM broadcast, red - equalized XM broadcast, black - the CD rip.
The three spectra are from three different tunes but they are tipical, not specific.
As to the difference vs the CD spectrum, my choice turned out not to be quite suitable. The sound has too many clipping which could explain the difference of the low frequency range.
Doing equalization, my goal is not to reconstruct the original but to change the spectrum according to my understanding of what it should be.

Finally, after having thought again, I really thing, the radio use three subbands Quadrature Mirror Filter banks.
And the technology at the sender side looks to me like:
- the Spectral Band Replication, thanks,  encoder prepares the high frequency range (above 15 kHz) first,
- the sound is Lowpass filtered at 15 kHz and then decimated to 30 kHz (33 kHz is less likely),
- then processed through three subband QMF bank where every subband is decimated at 10 (11) kHz and is encoded separately - which means the higest subband bit rate of 65 kbps is quite possible.