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Adding Speex Codecs to Barix Units

Folks-

I'm looking at the Barix units for remote broadcasts (voice, not music). I'm wondering whether I can add additional codecs to the units. Specifically, can I add the open-source Speex codecs? The Speex codecs are used in VoIP applications and have excellent quality at lower bitrates. I realize the new Barix 1000 units have VoiP codecs, but I'd like to modify the 100's if possible.

Thanks...
 
Maybe

Maybe. The Barix Instreamer and Exstreamers are programmed with Linux. Source code for their firmware is available on the web site. That said, you must understand that the Instreamer and Exstreamer 100s are designed to be value units-that is they are manufactured to do an excellent job of moving hi fi audio from point A to point B for an inexpensive price. Could Barix offer all the options that all you seem to want? Yes. Could they offer 24/7/365 technical service? Yes. BUT for them to do so, they would have to dramatically increase the price! Instead, they feel that the market segment they serve works for them. And believe me, it does! How much?

There is a Barix Exstreamer in every Whole Foods Market in North America. Soon there will be a Barix Exstreamer in every Sonic Drive In restaurant in North America. Clear Channel Satellite orders them by the HUNDREDS (thanks guys!). They are in use by stock brokers, local, state and federal courts, colleges, zoos, airports, race tracks, swap meets, more small (and not so small) radio networks then I can remember. Phoenix uses them to monitor airport noise, several states use them to feed their TIS transmitters, over a dozen Part 15 guys use them to feed multiple transmitters. NASA plans to begin using them. Even Leo Laporte (The tech guy) uses them.

Why are they so successful? Again, it's because they do a job very well, for a great price. That's what they are designed to do!

If you want low latency, try uLaw with Raw UDP transport.
 
Hey L.A. guy. I have a quick question for you since you brought up Sonic's radio network. Where are they studioed out of? Are the in downtown OKC or are the elesewhere as a provided service from another company? Also, what does CC Sat. Services use the Barix units for? Is it for distribution or to get audio to the for re-distribution over the bird?
 
OKCRadioGuy said:
Hey L.A. guy. I have a quick question for you since you brought up Sonic's radio network. Where are they studioed out of? Are the in downtown OKC or are the elesewhere as a provided service from another company? Also, what does CC Sat. Services use the Barix units for? Is it for distribution or to get audio to the for re-distribution over the bird?

With regards to Sonic, it is a provided service that is regionalized. I believe CC uses them for distribution to stations.
 
Hey LA guy, I sent you a private message with a couple questions on the Barix units a month back. Have you checked your messages here?
-D
 
dtube1 said:
Hey LA guy, I sent you a private message with a couple questions on the Barix units a month back. Have you checked your messages here?
-D

Best way to get hold of me is: dana(at)barix.com
 
Re: Maybe

LA_Guy said:
That said, you must understand that the Instreamer and Exstreamer 100s are designed to be value units-that is they are manufactured to do an excellent job of moving hi fi audio from point A to point B for an inexpensive price.

I know you are big proponent of Barix boxes, but I would refrain from classifying these as hi-fi... Of course this would depend on your definition of hi-fi, but in my opinion they are not it.


Regards,
Goran Tomas
 
Re: Maybe

Goran Tomas said:
LA_Guy said:
That said, you must understand that the Instreamer and Exstreamer 100s are designed to be value units-that is they are manufactured to do an excellent job of moving hi fi audio from point A to point B for an inexpensive price.

I know you are big proponent of Barix boxes, but I would refrain from classifying these as hi-fi... Of course this would depend on your definition of hi-fi, but in my opinion they are not it.


Regards,
Goran Tomas

A 196 kbps MP3 VBR encoded file ia so darn close to CD quality that 95 out of 100 users would not be able to tell the difference (196 kbps VBR coding is about the same as 256 kbps CBR). The other 5 listeners would not think it sounded 'bad'. It is FAR BETTER then any FM station! can transmit!

I believe the fidelity of an unprocessed high bit rate MP3 file will blow away the average audio that comes out of ANY digital audio processor, including any that you have designed.
 
Re: Maybe

LA_Guy said:
A 196 kbps MP3 VBR encoded file ia so darn close to CD quality that 95 out of 100 users would not be able to tell the difference (196 kbps VBR coding is about the same as 256 kbps CBR). The other 5 listeners would not think it sounded 'bad'. It is FAR BETTER then any FM station! can transmit!

I believe the fidelity of an unprocessed high bit rate MP3 file will blow away the average audio that comes out of ANY digital audio processor, including any that you have designed.

Please don't feel attacked. I'm not picking on you specifically, it's just that the quality of radio broadcasts has thumbled down so much in the last years and it has a lot to do with people taking perceptual coding easily and making assumptions and wrong conclusions. As someone who loves radio I'd like to see this trend reversed, hence my reply.

Btw, it would never occur to me to claim that FM processing is high fidelity. Processing is used to conform to some technical requirements of FM transmission and achieve some artistic goals, so by design it's not meant to preserve fidelity. We do however offer a high fidelity option in our processors - it's called the bypass preset! ;)

Likewise, I feel compelled to react when people call other things high fidelity that are not. Now I'm not against perceptual coding. Often times it lets us achieve higher quality audio that would be possible by not using it. But it should be treated with caution (maybe we should have a label for codecs like on the cigarettes ;) ) In my opinion, people should think twice before putting compression anywhere in their broadcast chain! They should consider the whole picture and carefully choose parameters such as bitrate and type of coding.

I agree with you that a 192kbps MP3 file encoded with something like lame will be very hard to distinguish from the original. What I don't agree with you are the conclusions that you draw from there. All codecs are not created equal, even if they use the same format. There are listening test out there which clearly show the differences between different off-line codec implementations for the same format. And then there are differences between off-line and real-time implementations. An MP3 file created on your computer is coded with an off-line codec, perhaps the one that will do two-pass coding can sound VERY different at the same target bitrate compared to a codec that does real-time coding. You should not generalize that if an MP3 file on your computer sounds good, a certain hardware codec will sound the same.

In particular, the Barix units in my opinion sound fairly bad. I heard a lot of positive comments about the Barix units and people saying it sounds good before actually having and opportunity to listen to the units. And when I did I was quite disappointed. At the highest quality settings which should do around 160 kbps (if not a little more than that) I wouldn't have expected to hear lot of artifacts, but in fact there are quite obvious artifacts going on. Like I said, in my opinion this is not high fidelity.

And you didn't give any consideration of the big picture. Is your music library compressed or not? Will you have cascading codecs? Where is your FM processing in the chain. Is it before or after the codec? Just this would require a whole article to explain the caveats...

What I'm afraid with comments like this is that people tend to get comfortable using coding and then the quality goes down rapidly. If you say it's great for STL, someone else will say "if it's good for STL, it's good for my music library". Even worse some will conclude if it's good for STL and sounds great, than it's good for my music library as well. Or - if I have coding in STL and it sounds great, I don't need to have higher quality in my music library. And they would all be wrong conclusions!

There's a use and benefit for perceptual coding in broadcasting, but you should not employ one easily and without taking everything into consideration. You should minimize the number of perceptual coding in your broadcast chain (from that field recorder all the way to the transmitter!) to a minimum - and that would mean 1. You should use the highest bitrate possible - very important! And finally, use the highest quality codec you can (MP3 is definitely not the greatest codec out there).


Regards,
Goran Tomas
 
Goran,

thanks for opening up the discussion of some very important issue. A few additional comments:

Since a processor will be in the chain when a codec is used for an STL the discussion must include how does the codec sound with processing before or after it?

Since digital bandwidth is getting cheaper, what are the options to reduce compression ratio? For example MP3 cannot go to a higher bit rate than 390 kbps.

What is the codecs performance of the codec in question across types of materials? MP3 has widely varying performance across different types of material.

And the possibility for cascading must always be considered.

Perceptual coders get most of there coding gain from destructive processes - removal of spectral content that is judged to be inaudible by the human perceptual system (though second generation and later algorithms such as MP3 and AAC also use entropy reduction techniques (such as Huffman coding). For example Layer 2 at a 4:1 compression ratio discards nearly 75% of the raw digital information.

This is in stark contrast to ADPCM coding techniques such as Enhanced apt-X, where under 5% of the raw information is lost at the same compression ratio. The newest algorithm in the apt-X family, apt-X Live adds Huffman coding to double the previous compression ratios possible.

Of course apt-X is highly regarded by picky post-production and other picky users including George Lucas. But not only is it true high fidelity, but it is highly resistant to multiple coding passes and, because it leaves more information intact, cascades well too.

All of this concurs with Goran's conclusions. Simple determination of "fidelity" at one particular stage is just the tip of the iceberg with regards to making deployment decisions.

As with processors, it really helps to listen to your own material through the system to see how the various factors play out with your material (including out of house material, spots, remotes, etc).

Inquiring minds will be rewarded in the end.

Rolf Taylor

Applications/Support Engineer

APT North America

www.aptx.com
 
Goran,

Thanks for the very well thought out response! Folks, this is something VERY important to keep in mind!

Just got back from the big apple...nothing worse than tuning into some of the biggest stations in the market, and listening to coded music libraries on-air with artifacts LOUD and PROUD after the processing got done with it all!

EEEWWWWW!!!! This? In US radio market #1 !!??!?!

-C
 
I've heard good things about AAC+, but has anyone tried it in broadcast applications? Do those Comrex IP remote units use it? The only problem I see with AAC+ right now is that the bitrate is a max of 128kbps 44.1 stereo and I would expect it to go higher for broadcast quality rates...

If you want to talk about high bandwidth STL coding, google OGG-FLAC. So far it is in the early testing stages but apparently VLC plays it if streamed to an Icecast server. Using mono coding FLAC gets around 640kbps that I've seen and depending on the stereo complexity I see about 950kbps for two channels. And remember its lossless so its just like sending PCM data! :)
 
RolfTaylor said:
Since a processor will be in the chain when a codec is used for an STL the discussion must include how does the codec sound with processing before or after it?

Indeed, there's a whole set of things to take into account there. In my previous reply I realized elaborating this would take a lot of space, so I decided it would be nice to write an article about it (as soon as I can find some free time do it!).

This is in stark contrast to ADPCM coding techniques such as Enhanced apt-X, where under 5% of the raw information is lost at the same compression ratio. The newest algorithm in the apt-X family, apt-X Live adds Huffman coding to double the previous compression ratios possible.

If I was building an STL and there was no way what so ever to go uncompressed, my first choice of compressed options would indeed be apt-x. I like that it is low compression ratio/high quality coding, I like the low latency and I like the broadcast quality and reliability of APT boxes.


Regards,
Goran Tomas
 
cgould said:
Just got back from the big apple...nothing worse than tuning into some of the biggest stations in the market, and listening to coded music libraries on-air with artifacts LOUD and PROUD after the processing got done with it all!

Hopefully we can educate people about this.

On the recent AES convention, there was a HUGE discrepancy about how people perceive perceptual coding depending on which room you went in. Experts who do perceptual coding research, design and blind comparative listening tests, were much more critical about perceptual coding than were people who use it in practice, including a lot of broadcasters. Some of the latter ones were throwing such comments out, that it was apparent they are not aware of "the dark side" of perceptual coding ;)

As with any tool, you can learn to use it properly or you can misuse it. The results will be quite different.


Regards,
Goran Tomas
 
gunterm said:
I've heard good things about AAC+, but has anyone tried it in broadcast applications? Do those Comrex IP remote units use it? The only problem I see with AAC+ right now is that the bitrate is a max of 128kbps 44.1 stereo and I would expect it to go higher for broadcast quality rates...

The HE-AAC codec (and the whole SBR family of codecs) is designed for very low bitrates. As such, it is not appropriate to be used on bitrates at or higher than 96kbps. You should use AAC for these higher bitrates as, in my opinion, it will sound much better.

Therefore you will typically use HE-AAC for dial-up speeds, low-bandwidth and/or congested networks, mobile networks, etc.

If you want to talk about high bandwidth STL coding, google OGG-FLAC. So far it is in the early testing stages but apparently VLC plays it if streamed to an Icecast server. Using mono coding FLAC gets around 640kbps that I've seen and depending on the stereo complexity I see about 950kbps for two channels. And remember its lossless so its just like sending PCM data! :)

Lossless is definitely something to consider. But I think it's application is more beneficial for file encoding, rather than transmission. It would increase initial STL costs, total latency and there's probably not much or any savings to be made if you have a ~1Mpbs link vs 1.5Mpbs (eg. E1/T1) link for uncompressed audio. Maybe I'm wrong...


Regards,
Goran Tomas
 
gunterm said:
I've heard good things about AAC+, but has anyone tried it in broadcast applications? Do those Comrex IP remote units use it? The only problem I see with AAC+ right now is that the bitrate is a max of 128kbps 44.1 stereo and I would expect it to go higher for broadcast quality rates...

AAC+ (also called HE-AAC) is used in the Comrex ACCESS. As noted here, it's a bit-reduced version of AAC, which is the gold standard in modern audio coding. It attempts to sound as good as AAC as a lower bit rate, so it's not popular in applications that have gobs of bandwidth. It does have a significant advantage on the public Internet, especially in challenging environments like 3G cellular, since a smaller stream usually results in higher stability.

The "killer" algorithm in the ACCESS is actually AAC-ELD, which is brand new and combines the aspects of AAC+ with AAC-LD. AAC-LD is a variant that has extremely low coding delay (around 50mS end-end).

Tom Hartnett
Technical Director
Comrex
 
comrex said:
gunterm said:
I've heard good things about AAC+, but has anyone tried it in broadcast applications? Do those Comrex IP remote units use it? The only problem I see with AAC+ right now is that the bitrate is a max of 128kbps 44.1 stereo and I would expect it to go higher for broadcast quality rates...

AAC+ (also called HE-AAC) is used in the Comrex ACCESS. As noted here, it's a bit-reduced version of AAC, which is the gold standard in modern audio coding. It attempts to sound as good as AAC as a lower bit rate, so it's not popular in applications that have gobs of bandwidth. It does have a significant advantage on the public Internet, especially in challenging environments like 3G cellular, since a smaller stream usually results in higher stability.

The "killer" algorithm in the ACCESS is actually AAC-ELD, which is brand new and combines the aspects of AAC+ with AAC-LD. AAC-LD is a variant that has extremely low coding delay (around 50mS end-end).

Tom Hartnett
Technical Director
Comrex

Plus I'll bet the ACCESS is a smart unit and can adapt based on the current network conditions to avoid skipping at the studio end.
 
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