ATSC Digital Audio Quality, Technology, Issues

[bentoronto]

Very nice of you to elaborate, even if many aspects of the information are wasted on me.

I'd like to know what the audio bandwidth is, the s/n in the hmdi plug coming out of my TV, and how much harmonic distortion, all relative to, say, the audio output of the CD carrying the original signal.

If the channel can not be specified in those terms, I wonder why? How about some typical empirical results?

Something less than amusing to hear JamesK say something less than perceptive like "Well buddy, it's digital so it can have no shortcomings," if I may paraphrase his post.

BTW, JamesK, there's no such thing as "Digital audio" only "audio transmission with part being modulated digitally." And yes, LOTS of places where the sound can get damaged between the CD player and my power amp, just to stick to electronic damage.

For FM, the quality specs are the required or federal regulatory specs of the broadcast process, like stereo separation, bandwidth, etc. Nothing like that with alternate channel sound?

[majortom]

Possibly, reading up on Dolby AC3 is what ur lookin to do, Wikipedia might have something...
Personally speaking, I think of WNED as a "class act" when it comes to the technical aspects
of their broadcast.
Doubt ya have much to be concerned with plugging it in to ur home stereo.

[Tom.F.1]

I did some searching, found a description on ADI's website.
http://www.adinternational.org/sap.html
Technical Information Regarding the Description Channel on Digital TV
The Advanced Television Systems Committee (ATSC) is an international non-profit organization which develops voluntary standards for digital television (DTV). Their publication, ATSC Digital Television Standard Part 5 — AC-3 Audio System Characteristics, defines the audio channel standards for DTV. The Standard, known as AC-3, specifies eight audio channels, one of which is known as VI (Visually Impaired), technically defined as service type 2 (in the range 0-7, with 0 representing the main audio signal). It further defines a "dynamic range control signal" this channel can use to "turn down" the main channel audio temporarily so description can be heard better. The channel may contain solely description, or the main program material can also be included. It is anticipated that the channel will be used for a merged soundtrack (that is, main program plus description, just like it is today on SAP).

While the FCC has endorsed this standard, the fact remains that many broadcast engineers are unaware of it, and there is no "practical" agreement on its use by broadcasters or implementation in TVs yet. (Thanks to Larry Goldberg of WGBH for supplying information regarding the AC-3 Standard.)

I have come accross audio only subchannels, and it is my understanding they can be multi program. Still looking for info on that.

[Tom.F.1]

Data, radio and non-broadcast signals
In rare cases, digital TV broadcasters have included the audio of a commonly-owned broadcast radio station among their subchannel offerings (for instance, KCSM-TV in San Mateo, California broadcasts the KCSM-FM radio station on its DT3 signal). Non-broadcast content, subscription TV channels or datacasting operations unrelated to the main TV programming are also permitted by the ATSC standard but are less commonly used.

http://en.wikipedia.org/wiki/Digital_subchannel

[bentoronto]

Thank you much Tom.F.1. and majortom.

My interest in audio performance specs is the natural inclination of audio hobbyists who thrive on perfecting concert-hall reproduction (that is an undefinable fantasy, of course) and, at least for the techies, want objective measurements as far as possible.

But that is very different than the world of satisfying illusions natural to home theatre, ipods, and the like. No moral judgment intended.

Which brings me to Dolby: are there straightforward ways to evaluate systems which include Dolby processing, signal compression, mpeg, and other elastic or trade-offable parameters? My guess is that at least not in the traditional terms audio hobbyists seek.

So my conclusion is that nobody much cares to measure or to regulate the quality of these alternate audio channels, their quality can vary from place to place and time to time. At best, they may be capable of very high quality and multi-channel transmission, exceeding FM broadcasting.

Aren't there broadcast equipment manufacturers who feature audio specs for their gear?

[stampeder]

While audio subchannels are not a technical problem under the ATSC standard, there has been a great deal of money, time, and effort spent in the U.S. on digital HD Radio to improve the FM Radio band, so that's the show-stopper.

The political and economic reality of the U.S. broadcasting industry, in which outside entities are openly coveting OTA TV spectrum, is such that the TV station owners don't want to use their allotted bandwidth for something that the Radio broadcasters (often the very same companies) are already achieving with HD Radio.

That's it, in a nutshell. Audio-only sub-channels do exist on rare occasions (usually on non-profit or listener-supported stations) but are generally a dead issue on conventional TV stations.

We'll avoid discussing HD Radio in this thread. The link above gives great threads for that.

[bentoronto]

As always, big thanks to stampeder. I read a lot of your links and maybe am more confused than ever. What a dogs' breakfast - big mess esp. in Canada.

Seems to me, if anybody is interested in ignorant musings, that getting broadcast and digital FM into the DTV band makes a lot of sense (esp. for the OTA crowd) and all the other systems have even worse obstacles.

Mostly, I think in terms of antennas being the weak link and so I think an approach where everybody uses a "UHF" antenna for all home reception (A and V) computes best. For audio-only rooms, there could be receivers that pick up "radio" stations and also the alternate subchannels on TV stations, both in a DTV band. For home-theatre rooms, you could listen to radio broadcasts on the alternate sub-channel on TV channels but also readily tune to radio-only channels assigned to spare spaces (and even listen to TV shows without the picture).

Best of all, and thinking here of the 1948 FM mess, FM broadcasting can continue indefinitely where it is and how it is.

[holl_ands]

Originally Posted by bentoronto

Thank you much Tom.F.1. and majortom.

My interest in audio performance specs is the natural inclination of audio hobbyists who thrive on perfecting concert-hall reproduction (that is an undefinable fantasy, of course) and, at least for the techies, want objective measurements as far as possible.

But that is very different than the world of satisfying illusions natural to home theatre, ipods, and the like. No moral judgment intended.

Which brings me to Dolby: are there straightforward ways to evaluate systems which include Dolby processing, signal compression, mpeg, and other elastic or trade-offable parameters? My guess is that at least not in the traditional terms audio hobbyists seek.

So my conclusion is that nobody much cares to measure or to regulate the quality of these alternate audio channels, their quality can vary from place to place and time to time. At best, they may be capable of very high quality and multi-channel transmission, exceeding FM broadcasting.

Aren't there broadcast equipment manufacturers who feature audio specs for their gear?

You're barking up the wrong tree....IT'S DOLBY DIGITAL (DD)!!!!! BitsOut=BitsIn.

The audio quality in the source program will be perfectly delivered from digital source to destination
(the Dolby Digital Decoder in your AVR).

The real questions to ask are 1) how good is DD and 2) what happened before it became DD?

Quality of DD, whether DD2.0 or DD5.1 can be researched elsewhere (do you need references?)....
I find it superior to CDs, but not quite as good as DTS, SACD, DVD-Audio, et. al.

The REAL problem is what did the FM or TV Broadcaster (or Sirius, or MusicChoice Cable supplier)
use as the SOURCE (cassette, CD, mp3, aac, etc. file) and what did they DO to it, if anything???
I'll bet that many of them are ripping CDs to either mp3 or aac files (YUCK!!!). The digital files
would be converted to analog audio input for an ATSC Encoder that converts to DD (DOUBLE YUCK!!!)

When digital compression algorithms are used in cascade, bad things can happen.
See Table 3 for what happens when dissimilar cellphones "talk", e.g. GSM to/fm CDMA:
http://www.ocecpr.org.cy/media/docum...04-2000_VI.pdf

BTW: Since ATSC is restricted to DD, SACD source could be "down converted" to DD5.1
(not quite as good as the SOURCE) and directly inserted for transmission.
PS: Although MPEG2 can carry up to 12(?) PCM audio streams or up to 48.12 aac streams,
there are no specs on how to provide say PCM7.1 sound to your AVR....and ATSC chose
DD5.1 instead of MPEG2 to carry sound...

[bentoronto]

The audio quality in the source program will be perfectly delivered from digital source to destination
(the Dolby Digital Decoder in your AVR).

Certain pieces of the chain of transmission have low "error" rates. But claims of system perfection in engineering textbooks need empirical confirmation.

[majortom]

When digital compression algorithms are used in cascade, bad things can happen.
See Table 3 for what happens when dissimilar cellphones "talk", e.g. GSM to/fm CDMA:
http://www.ocecpr.org.cy/media/docum...04-2000_VI.pdf

I wonder why they didn't include the GSM AMR Speech codecs?
I guess partly because the date of that paper was 2000.
The GSM Codecs they have listed there are GSM Speech Ver1, and Ver2 codecs. GSM AMR being GSM Speech Ver3 which is what is used today on GSM networks and handsets. Unless, of course ya had a really, really old phone, the GSM network is backward compatible with ur old codec phone.
Sorry, off topic again...

[iblackford]

I'm actually quite interested to hear what reasons you have for thinking that the digital data stream could not be delivered perfectly. Engineers employ many methods to ensure data and signal integrity (controlled impedance traces, pre & post equalization, lvds, etc...). The last defence against all of these things is an error-correction algorithm, which if it cannot fix the error will result in audio drop-outs.

Holl_ands is correct that the focus should be put more on pre-digitization to make sure that the audio was not mangled in any way.

[bentoronto]

In practice, no sweat moving digits around pretty well on most good days, generally, I'd guess, if not too far from the transmitter. But what about modulation/coding and demodulation into analogue signals, transmitters, limiting conditions, and a lots of other pieces inherent in the transmission chain and which you are ignoring?

[JamesK]

^^^^
Modern digital communications systems take all that into consideration. There are two big factors. One is binary. A bit can be heavily distored and still be recognized. This also means significant noise levels can be ignored. Then, allowing for the possibility of bad bits getting through, error detection and correction is used. On computer networks, the correction takes the form of retransmission of the error packets. In things like broadcast or CDs, forward error correction is used, in that sufficient redundant information is included to recover up to a certain level of errors. Another technique is to use interleaving of data, so that instead of one big hit, that might overwhelm the forward error correction, it becomes multiple small hits that can be corrected. There is a lot more to this, but essentially, error free transmission is practical.

[holl_ands]

I wish error free reception was guaranteed....but it's NOT.
My cable DVR can display the long term average for DETECTED MPEG2 bit error rate,
which NORMALLY runs in the 10-8 to 10-7 range....and will be glitchy at 10-6.
BUT, even on cable, video glitches are still very frequent and are NOT detected
in my MPEG2 error statistics summary....hence they are generated at the SOURCE!!!!

That's CABLE!!! OTA would be worse, esp with multipath dropouts.
Fortunately, MPEG2 has an inherent bit error CORRECTION code
that fixes bit errors when they are at a very low ber.....but is
overwhelmed when a multipath fade occurs and you see a video glitch.
Also fortunately, the audio signal is also protected against fading and many
ATSC decoders go the extra step to "fill-in" the audio waveform when missing
packets occur, "hiding" the (very small) audio defect so you usually can't hear it....

[bentoronto]

JamesK writes

Modern digital communications systems take all that into consideration. There are two big factors. One is binary. A bit can be heavily distored and still be recognized. This also means significant noise levels can be ignored. Then, allowing for the possibility of bad bits getting through, error detection and correction is used. On computer networks, the correction takes the form of retransmission of the error packets. In things like broadcast or CDs, forward error correction is used, in that sufficient redundant information is included to recover up to a certain level of errors. Another technique is to use interleaving of data, so that instead of one big hit, that might overwhelm the forward error correction, it becomes multiple small hits that can be corrected. There is a lot more to this, but essentially, error free transmission is practical.

In trying to justify your position that "digital" means a system is without errors, am I mistaken that you are confabulating together in one paragraph broadcasting with digital systems in which the receiver can call for a re-transmission from the sender (definitely not relevant to broadcasting) and systems where there's leisure for recalculation and adjustment (that's not relevant to broadcasting)?