Scott uses MP2...Why?

[Randy Lahey]

Why does Scott use Mp2 and not mp3? Everything we receive trough email and ftp is ALWAYS mp3, so why make us go through the extra step of conversion?

If it's a sound quality issue (which I don't think it is but I could be mistaken) we obviously aren't "gaining" any quality.

Is it a copyright issue? I know of a DAW that can't convert to mp3 because they didn't pay for the rights (Cubase I think).

Can other automation systems use mp3?

Randy.

 

[RadioMan203]

I can't speak for Scott, but when I was running the DOS-based Digital Jukebox it was because the Antex audio cards had on-board codecs for mp2.

 

[OttoM]

Bill's on the right track here.

The early (1990's) Scott-branded ASI cards had onboard MPEG-2 decoders and won't play MP3s. The no-MP3 rule is probably to keep compatibility with the older hardware.

Many other systems can use MP3 - but in this day and age of cheap hard drives (~$75 for 1Tb) there's no reason to run ANY compression on an automation system. Convert everything to WAV and be done with it.

 

[boiseengineer]

No reason to use MP2 on the Scott. Record/TLC audio in as linear wav.

 

[Sgeirk]

I've heard others echo this, and it's also my opinion that for broadcast MP2 produces fewer artifacts when heavily processed, although I'm 100% certain that has nothing to do with their decision to use MP2. That said, we don't compress anything in Scott.

If you have a bunch of mp3's to convert, use another program to decode them in bulk like CDex or dbpoweramp, it can save a little time, especially if you're putting them in as wav's.

 

[radiorob2.0]

Bill's on the right track here.

The early (1990's) Scott-branded ASI cards had onboard MPEG-2 decoders and won't play MP3s. The no-MP3 rule is probably to keep compatibility with the older hardware.

Many other systems can use MP3 - but in this day and age of cheap hard drives (~$75 for 1Tb) there's no reason to run ANY compression on an automation system. Convert everything to WAV and be done with it.

It's not worth converting an mp3 to wav because the loss doesn't magically return, it's just becomes a bigger file. But I agree that with the price of hard drives wav files are the way to go from the start.

 

[NoTimeForSleep]

I can't speak for Scott, but when I was running the DOS-based Digital Jukebox it was because the Antex audio cards had on-board codecs for mp2.

Many older systems used MP2 with older ISA bus audio cards. It was less lossy then using MP3 and was necessary due to the limited hard drive sizes available back then.

 

[ChiefOperator]

mp2 was adopted as the broadcast standard in the days before mp3 gained popularity. Also, ASI cards used mp2 codecs. I also believe, as someone mentioned, that mp2 is a "smoother" codec and doesn't have the artifacts that mp3 does (especially when passed through a processor). Our music library is encoded at mp2 (384kbps), which is the first layer of compression (3:1, I believe). Our processor likes this format and the sound is quite good.

Today, there is little reason to use compression, given the size and cost of HDs. Go PCM....

 

[Randy Lahey]

Thanks guys. I appreciate all of the responses.

What about other automation systems? Specifically Google Automation?

Can they play mp3?

 

[TomT]

Simian will handle MP3, before our ops mgr. graduated he would cut weather forecast, school closings in his apartment at the university and transfer them directly into the program using go to my pc.

MP3 does require separate licensing, the Fraunhofer Institute holds patents on it. They collect directly from the software manufacturer.

 

[RadioMan203]

MP3 does require separate licensing, the Fraunhofer Institute holds patents on it. They collect directly from the software manufacturer.

I know some software uses the LAME codec, which is suppose to be compatible with mp3. I don't remember all the details of it but I thought it was suppose to be an open source codec.

 

[Sgeirk]

Hey Radiorob:

It takes less time to convert a wav file to the scott mp2 format than it does an mp3 to their mp2 format (seems more than 2x-4x as long), which is why I suggested using an outboard program to convert to wav first. Especially if you have a stack of mp3's that have to be thrown in.

I mean, get coffee, use the bathroom, go make a phonecall, dub other stuff in the system...I'm just suggesting using something better for the task at hand.

 

[rocketgal]

Please, for the sake of your listeners (and our industry) , don't convert an MP3 to MP2!!!

MP3 files by themselves really aren't good enough to broadcast IMHO, especially if you're looking at subsequently using a compressed STL or HD Radio. Personally, I think lossy compression adds to listener fatigue which reduces time spent listening. If you cat be uncompressed, do it. If you can't, use as high a bit rate as possible.

Rox

 

[syvjeff]

Scott isn't the only automation that uses MP2, because of consolidation of groups, music libraries and the size of data needed to be moved in the year 1997 MP2 became a standard before MP3 was really available. Many CC stations still use MP2 in their Prophet systems.

 

[RolfTaylor]

MP2 is a perfectly god codec as long as you don't try to compress more than about 4-5:1.

16 bits/byte x SampleRate Hz x 2 channels /Y = bit rate , where y is the compression ratio

So, depending on the

16 x 44100 x 2 /4 = 325800 bits per second
or
16 x 48000 x 2 /4 = 384000 bits per second
or
16 x 32000 x 2/4 = 256000 bit per second

At these bit rates Layer 2 is better than layer three. In fact that is why Layer 3 only goes up to 330kbps.

Then again, if you only need 4:1 compression ratio enhanced apt-X is even better since it is much less lossy than layer 2 and therefore cascades with downstream compression (if any) extremely well.

So, to summarize, Layer 2 is a perfectly good tool when used within its limitations. MP3 became popular in broadcast because it could do good to excellent quality stereo at 128 (e.g. a single BRI ISDN circuit). That is about 10:1

One thing about MP3, is that more source dependent than the later codecs. Its MOS score varies considerably across the different test items. AAC is good for higher ratios of up to 15:1 with better source to source consistency.

The original apt-X had problems with certain transients but this problem has been solved with enhanced apt-X.

Rolf Taylor
Applications/Support Engineer
APT - an Audemat Company

 

[ChiefOperator]

Yes, we've had excellent results with our music library which is encoded in mp2 @ 384kbps.

Rolf, how would one convert into enhanced apt-x? I've not seen any audio "file" that is enhanced apt-x, yes?

 

[gtomas]

So, to summarize, Layer 2 is a perfectly good tool when used within its limitations. MP3 became popular in broadcast because it could do good to excellent quality stereo at 128 (e.g. a single BRI ISDN circuit). That is about 10:1

Layer III was indeed better at 128 kpbs than Layer II and hence its popularity in broadcast codecs and links that used ISDN (which had a maximum throughput of 128 kbps).

However, as some people might misinterpret your words, I'd like to point out that I wouldn't call MP3 file encoded at 128 kbps to have an excellent quality. MP3 needs at least 192 kbps, preferably 256 kbps to sound excellent, IOW indistinguishable from the original. Of course, there are a lots of buts in this, mainly the quality of the encoder, but even with the best encoders the quality of MP3 at 128 kbps might be very good, but not excellent.

On the other hand, AAC was found to be transparent at 128 kbps by the EBU and though I would still use a higher bitrate as I can still hear some artifacts at 128 kbps, it is a much better codec than MP3. Too bad it wasn't available in the early days of ISDN and too bad it had a hard time getting attention with the popularity of MP3. Regarding AAC and bitrates, I applaud the Apple which set the standard of the songs purchased through iTunes at AAC at 256 kbps. This is undeniably excellent quality and no-one can object to it for personal listening.

However, in these days (and I guess even the birds on the roof are singing this by now) I would avoid using any coding in the broadcast facility. If there's really, really, no other possibility, I would limit it to only a single pass of perceptual coding. For example a link. Or in the case of digital broadcast, the final HD Radio/DAB/streaming encoding.

The choice of the codec in these scenarios is important, as there are better quality and better efficiency codecs. However, perhaps the most important factor in perceptual coding is the bitrate. Whichever codec you use, use the highest bitrate possible! At high bitrates such as 320 kbps, Layer II, Layer III and AAC will all sound excellent. Again, if you really can't get away from using coding at all...


Regards,
Goran Tomas

 

[gtomas]

Yes, we've had excellent results with our music library which is encoded in mp2 @ 384kbps.

Rolf, how would one convert into enhanced apt-x? I've not seen any audio "file" that is enhanced apt-x, yes?

I believe Rolf was talking about real-time coding in the STLs and remote broadcast links, rather than off-line music library/file encoding....


Regards,
Goran Tomas

 

[ChiefOperator]

Thanks Goran and Rolf for the explanations. Goran, I too can notice a difference between mp3@128 and mp3@192.

Just for info, I believe the new MAX receivers are using the AAC codec, at least that is what WW1 has told me.

 

[RolfTaylor]

Goran,

as usual, you bring up an important point:

I will make the assertion that MP3 can be excellent if the following criteria are met:

--Encode at 32 kHz sample rate: I realize we could get started on a whole thread about 15 vs 20 kHz, but in this case the 15 kHz will sound better because we've reduced the compression ration
--Use Joint stereo. The MP2 vs MP3 Joint stereo modes are quite different.

Now, lets look at the compression ratio's involved as I just said "about 10:1" in my original post.

Solve the equation I gave before for compression ratio gives us 16 bits * SampleRate* 2 channels/bit rate = Ratio

16bits * 32,000samples per second * 2/128000bits per sec = 8:1

16 * 44,100 * 2/128000 = 11:1

16 * 48,000 * 2 /12 = 12:1

I invite anyone interested in learning to hear coding artifacts to try Layer 3 at 128 with 32,000 kHz and 48 kHz sample rates and you will definitely be able to hear a difference.

At 48,000 sample rate you will clearly be able to hear the difference between Stereo and Joint-Stereo, with the joint stereo overall being much better.

While you are at it, try Layer 2 at 384 kbps and you will find it sounds excellent, even at 48 kHz sample rate.

Next re-read Goran's post because I agree with his precautions and warnings!

Rolf Taylor

So, to summarize, Layer 2 is a perfectly good tool when used within its limitations. MP3 became popular in broadcast because it could do good to excellent quality stereo at 128 (e.g. a single BRI ISDN circuit). That is about 10:1

Layer III was indeed better at 128 kpbs than Layer II and hence its popularity in broadcast codecs and links that used ISDN (which had a maximum throughput of 128 kbps).

However, as some people might misinterpret your words, I'd like to point out that I wouldn't call MP3 file encoded at 128 kbps to have an excellent quality. MP3 needs at least 192 kbps, preferably 256 kbps to sound excellent, IOW indistinguishable from the original. Of course, there are a lots of buts in this, mainly the quality of the encoder, but even with the best encoders the quality of MP3 at 128 kbps might be very good, but not excellent.
<SNIP>

Regards,
Goran Tomas