Why does it sound different on the Radio...

[Marcel]

A post by a broadcasting tech friend on FB, along with recent paid work I've been doing at a local radio station has prompted me to write this slightly informative piece................ and maybe start a discussion...

Something which baffles many yet is standard practice for the professional broadcast industry is the use of specialised audio processing on EVERYTHING that is broadcast.

In the early days of broadcasting (back when AM radio was King) it was found that over driving transmitters was a very bad thing which apart from adding unwanted distortion to the voice of the DJ also had the problem of making it very difficult for the listener to tune in their often quite basic radio. Other problems included 'splatter' where one radio station could be heard on multiple places on the old school dial. So to alleviate this it was decided and internationally legislated way back in great grandads day that hard audio limiters in the audio chain would be mandatory and ALC (Automatic Leveling Control) would be an recommended option that each station could implement at their discretion.

Fast forward to the '70's and the whole broadcast audio limiting/ALC changed into a race into who could sound the best/loudest. A plethora of new solid state devices that optimised the audio for transmission were devised and implemented. This all came under the thinking that a smaller and cheaper to operate transmitter if modulated correctly could sound as loud as a transmitter 10 times the size and cover as many miles or kilometres as one twice the size, so it was a cheap way of boosting presence on the dial, to sound bigger, better, more awesome all for less money....

FM radio and TV didn't escape the madness. They too have their own versions driven by the desire to sound the best/loudest while staying within the national Federal broadcasting regulations.

So what do they do?. Well essentially the hard limiter by legislation must be there, often in the form of a fast attack and fast release 20:1 compressor with the threshold set at 95% modulation as the last element just before being sent to the transmitter, purely to prevent over modulation. But the ALC part is where it gets interesting. The automatic levelling part is for the most part again is a 20:1 compressor that has a fast attack and very very slow release and a threshold set quite low in the 10 to 30dB below 0VU range used with the intention of averaging all the audio to a predetermined volume. As part of the averaging the audio is broken down into frequency ranges that are also averaged. The number of frequency groups is determined by the type/style of typical program material, but typical breakdowns are usually between 4 and 8 frequency groups. If you think of a 2way or 3way crossover in your typical loud speaker it's a similar type breakup of frequencies being done, one or two bass frequency groups and one or two mid frequencies and one or two treble frequency groups. All the individual frequency groups are ALC'd individually before being recombined into the one signal that is sent to the hard limiter and then the transmitter....

Setting up one of these 'audio processors' is a black art reserved for the station technical engineer under the verbal direction of the station management team. Nobody 'plays' or fiddles with the station audio processor as it does invariably affect how the station sounds to all its listeners.

So if you were ever wondering why your CD or MP3 sounds so vastly different on the radio than from your home stereo, that is why.... The processor does change the 'sound' of the audio passing through it...

If you're interested in seeing a very popular model of these 'audio processors' in operation then follow the link which will take you to a YouTube list of demonstrations.... https://www.youtube.com/results?sear...=orban+optimod

The photos are of the one I had the joy of adjusting just the other day...

[blinddrew]

So basically it's got a heavy multiband compressor on the end that squashing the snot out of everything?

[Marcel]

So basically it's got a heavy multiband compressor on the end that squashing the snot out of everything?

Yeah, some see it that way. Most see it as a way of optimising the limited capabilities of the 'transmission medium'... an effort to stay as close to 95% as is reasonably possible.

Back in the '70's and early '80's there were some stations that 'squashed the snot' out of everything. While popular for a while it did bring about the condition known as 'listener fatigue', and due to the loss of listeners that the condition brings with it most stations eased back on the 'squash everything' aspect in favour of a 'consistent easy listening all day' approach. If you were to A/B the two settings the difference (to your ears) is obvious...

[dave.king1]

The Orban loudness meter is one of the mixing tools I use from time to time, not a huge fan of it for my needs though.

Multiband compression and master limiting are the major magic ingredients in the loudness war from a recording perspective

[wazkelly]

Good read Marcel.

Can you also enlighten us on the reasons why Vinyl compared to CD compared to MP3 sound so different?

From an uneducated perspective MP3's all sound tinny with lots of tops and minimal bottom end, probably done that way to prevent blowing your brains out through those tiny earbuds. CD's compared to vinyl sound bright but lack the analogue warmth. For me vinyl is the valve amps equivalent if that makes sense?

Interested to hear your thoughts.

[Marcel]

Good read Marcel.

Can you also enlighten us on the reasons why Vinyl compared to CD compared to MP3 sound so different?

From an uneducated perspective MP3's all sound tinny with lots of tops and minimal bottom end, probably done that way to prevent blowing your brains out through those tiny earbuds. CD's compared to vinyl sound bright but lack the analogue warmth. For me vinyl is the valve amps equivalent if that makes sense?

Interested to hear your thoughts.

Vinyl & CD & MP3.... That to me is an unfair battle...

The absolute beauty of vinyl is in that it is entirely analogue. That minuscule little Diamond shaking back and forth as it tracks down the groove of your favourite 33rpm LP being mechanically coupled to a magnet surrounded by two just as tiny coils to generate a pair of electrical signals that are then amplified before going to another set of much bigger coils in much much bigger magnetic fields that ultimately generate those magical sounds we love.... Nothing is missed, All of it is there, including the dust and static that vinyl is known for...

CD is clean. 44100 times a second the incoming analogue audio is measured and allocated a value. A value that will never change for the rest of eternity. No matter how many people jump on that wooden dance floor if you put enough of those values into a big enough buffer the D to A conversion at the loud speaker will always be representative of the original recorded value. But some question the process, some 'Golden Ears' consider the process flawed, and possibly for good reason.... In my mind I see it this way. On a 440Hz note and with 44100 per second sample rate there will be close to 100 individual samples available to rebuild a single cycle of the original "sound" at the loud speaker, but on the 88th note of a piano at 4186.1Hz there will only be about 10 individual samples to rebuild all the complexities of that specific piano note. Ten samples is still quite a lot and will be fairly representative but it will not be 100% accurate. Thankfully though we don't have dust and static to contend with.

Then there is MP3.... the WIN ZIP of the music world. As CD is simply a series of numeric values, and if we jam those numbers into a computer and ZIP them then we can cram more and more music into less and less hard drive space. We can shrink the files more if we re-sample to a lower bit rate, which is the same as changing our original CD sample rate of 44100Hz to something less like 32000 or 24000 or 16000 or 9600Hz. We can even go to in between sample rates by using a bit of maths and predicting what a sample might be if it were taken somewhere between two 44100Hz samples and then using that newly mathematically created sample as a way of creating an entirely new sample rate... It's just numbers, and we have the calculator... But as we know everyone has a different ear for these things and to some people these fictitious sample rates are just wrong.... They sound bad. They might be okay for recording a telephone conversation or a Uni lecture on a 2GB SD card but in no way should they be used for quality music. Unfortunately many do, yet thankfully there are many more who don't.

Most radio stations these days do not have a vinyl or a CD collection... Nearly all have only MP3's. Most of those MP3's are at a reasonable bit rate. There will be a CD player somewhere in the studio used to add to the collection of MP3's, and usually one of the managers will have access to a vinyl to CD recorder, so if there is a 'Golden track' brought in they can find a way to add it to the master library all in the name of posterity ....

[Simon Barden]

In my mind I see it this way. On a 440Hz note and with 44100 per second sample rate there will be close to 100 individual samples available to rebuild a single cycle of the original "sound" at the loud speaker, but on the 88th note of a piano at 4186.1Hz there will only be about 10 individual samples to rebuild all the complexities of that specific piano note. Ten samples is still quite a lot and will be fairly representative but it will not be 100% accurate. Thankfully though we don't have dust and static to contend with.

As Doc Nomis said, the Nyquist Theorem states that if the sampling rate is more than double the maximum frequency you are trying to sample, then you can reproduce that sound perfectly. It doesn't matter if the frequency is 20 Hz or 20,000Hz, if you are sampling at 44.1kHz, then both frequencies get reproduced accurately. Each sample represents a point on a curve (not the height of a step) and the D/A converter recreates the curve as a continuous voltage wave.

There's a great video on this web page that explains almost all of the misconceptions about digital audio. https://xiph.org/video/vid2.shtml

[Marcel]

As Doc Nomis said, the Nyquist Theorem states that if the sampling rate is more than double the maximum frequency you are trying to sample, then you can reproduce that sound perfectly. It doesn't matter if the frequency is 20 Hz or 20,000Hz, if you are sampling at 44.1kHz, then both frequencies get reproduced accurately. Each sample represents a point on a curve (not the height of a step) and the D/A converter recreates the curve as a continuous voltage wave.

There's a great video on this web page that explains almost all of the misconceptions about digital audio. https://xiph.org/video/vid2.shtml

Yeah.... But !...

I 100% agree with the theory, The theory that for any given series of sampled points there is only one mathematically correct output waveform that any reasonable D to A converter should deliver. However in the real world we need to deal with the hardware that does have its limitations.

When testing with pure tones the limitations are almost non-existent. The prediction of the next sample point and the slope of the voltage change are very close to ideal so a smooth transition from one sample to the next is easily calculable, and thus noise outside of the frequencies of interest are just as easily minimised. Even slightly more complex two tone or three tone tests yield similar high quality results.

A 'foot to the floor' or a '100% difficulty' test that has not often been demonstrated is one where a wide band white or pink noise is passed through the A to D to A process and then compared to the identical delayed source signal on a scope with the inputs in differential mode. The line on differential mode scope screen will display all differences between the original signal and the ADA signal, and those differences will appear as noise on the line on the screen. The less noise the better the ADA process. Sadly there will always be some noise.

What the noise represents are the differences between the original signal and what is presented by the ADA process. How this noise affects our human perception of any given ADA signal when compared to a pure original analogue signal is a matter or the matter open for debate...

If it's a digital 99.999% or analogue 100% accurate there are 'golden ears' that claim they can hear it, however I suspect most are like me and hear like me so I doubt most would be unable to tell the difference.

A case in point - There is an Israeli communications company that wanted to cram as many phone users onto their E1 (2Mb) streams as possible, so they conducted tests where they reduced the bit rate of their phone circuits to find the minimum sampling rate where communications and ineligibility were maintained at just over 97% of the spoken word. Huge teams of people chatting for hours over low bit rate circuits and giving it a rating. Apparently a 16k data rate gives that 97%, so that means on a 64k data circuit that formerly was the international standard for one phone line they could cram 4 phone lines, or have 3 phone lines and have spare signalling capacity on a 56k data circuit feeding a T1 trunk. Their E1 trunks jumped from 32 callers up to fully served 128 clients, and the clients for the most part wouldn't know the difference. I've listened extensively to these 16k audio circuits and for a phone line of 300 to 3kHz range the ineligibility is still pretty darn good. However you'd be gob-smacked to know who else consistently now uses this bit rate trick for other purposes in today’s world and gets away with it, and mostly only to save money.

For me 44.1k is awesome and works just fine. Essentially flat 20Hz to 20kHz response, +90dB S/N... nil pops or crackles.... Awesome.!! .... Get rid of all those coupling caps and DC couple from D/A to speaker is the only way it can get better...

[DrNomis_44]

Good read Marcel.

Can you also enlighten us on the reasons why Vinyl compared to CD compared to MP3 sound so different?

From an uneducated perspective MP3's all sound tinny with lots of tops and minimal bottom end, probably done that way to prevent blowing your brains out through those tiny earbuds. CD's compared to vinyl sound bright but lack the analogue warmth. For me vinyl is the valve amps equivalent if that makes sense?

Interested to hear your thoughts.

The compression algorithm used in mp3 files is a lossy one, essentially what is being done is that a lot of the subtle parts of the music which we don't notice much, is stripped-off in order to reduce the amount of digital data, Marcel could probably explain it more clearly.

A typical audio CD (compact disc) has a dynamic range of about 105dB or so, that is, from the softest sound that can be heard to the loudest sound, there's a range of about 105dB, on the other hand, a vinyl disc has about a 30dB dynamic range, which is a lot less than a CD, reason being is that the fine grooves cut into the surface of the vinyl disc can only accommodate so much before the stylus starts jumping out of the groove (this is called skipping), the vinyl disc system also tends to produce a characteristic type of distortion which is similar to what the human ear naturally produces by itself, so as a result, a vinyl disc will tend to sound warmer than a CD even though the CD system is technically superior (supposedly).


In digital audio, there is something called the "Nyquist Theorem" which basically states that to accurately reproduce the highest frequency in the frequency response of a digital audio system, the sampling frequency has to be at least twice the highest frequency that is likely to be reproduced.

https://blog.telegeography.com/what-...does-it-matter


The thing to note is that there is no "perfect" audio media system, each one has it's good points and flaws, when CDs first became available to the paying public, they were regarded as perfect and indestructible, it wasn't too long after that when we started seeing the flaws that proved that CDs were far from indestructible, they were prone to skipping due to circumferential scratches on the surface of the CD, the digital encoding of the information on the CD also included an error-correction system that could cope with loss of data due to a 4mm hole drilled through the CD, or a radial scratch going from the centre of the CD outwards.

[dave.king1]

MP3 algorhythms are massively compromised compared to most other containers Waz.

If you have something of your own in various formats put it up on Soundcloud or YouTube and have a listen to how each is treated by that platform, I have put stuff on SC in both MP3 and 16bit wave format and it's hard to tell them apart, cut them both to Redbook and you'll hear the difference.

The difference between CD and vinyl is the brutal lack of dynamics zeros and ones compared to analogue, simplistic I know but not sure how to better describe it