Is anyone using anything higher than 128kbps/44.1mhz for spot production?
-
Get involved.
We want your input!
Apply for Membership and join the conversations about everything related to broadcasting.
After we receive your registration, a moderator will review it. After your registration is approved, you will be permitted to post.
If you use a disposable or false email address, your registration will be rejected.
After your membership is approved, please take a minute to tell us a little bit about yourself.
https://www.radiodiscussions.com/forums/introduce-yourself.1088/
Thanks in advance and have fun!
RadioDiscussions Administrators
You are using an out of date browser. It may not display this or other websites correctly.
You should upgrade or use an alternative browser.
You should upgrade or use an alternative browser.
Bit/Sample Rates
- Thread starter hungryhawk
- Start date
A
Auralous Sublimate
Guest
GOOOOOOD question.
My personal preference (and one I've found is common) is no lower than 192 (cd quality)-- certainly not lower than 160 (probably best for your needs).
Consider that at 128bps, the data compression is so much that your get that distinct internet "stringly" sound. Yuck.
Also consider that this spot will be data compressed once more as it is entered into whatever system for on-air playback you're using.
The file sizes aren't that different, but the boost in quality will make you sound so much better.
With that, anyone who reads this with a positive opinion of HD Radio should come to grips with the fact that-- unless everthing you play is off CD-- you'll NEVER give REAL "CD quality sound" to the consumer since putting it "into the system" compresses the data to imperfect. I'm just saying. Hell... we offer people a polished turd of a product... and can't even really give them THAT! Again, just saying.
My personal preference (and one I've found is common) is no lower than 192 (cd quality)-- certainly not lower than 160 (probably best for your needs).
Consider that at 128bps, the data compression is so much that your get that distinct internet "stringly" sound. Yuck.
Also consider that this spot will be data compressed once more as it is entered into whatever system for on-air playback you're using.
The file sizes aren't that different, but the boost in quality will make you sound so much better.
With that, anyone who reads this with a positive opinion of HD Radio should come to grips with the fact that-- unless everthing you play is off CD-- you'll NEVER give REAL "CD quality sound" to the consumer since putting it "into the system" compresses the data to imperfect. I'm just saying. Hell... we offer people a polished turd of a product... and can't even really give them THAT! Again, just saying.
320kbps. Period. I don't care for using mp3 at all, but sometimes it is needed, so everything is at 320kbps. Unless it happens to be a mono voice file, in which case it is 128kbps mono (about equal to 320kbps stereo). And that is only for delivery. Any in-house spots remain as wav files. And yes, with HD, you get double-masking, which is absolutely grotesque if you start with anything lower than 320kbps. A 128kbps mp3 sounds about like an 80kbps mp3 through HD. I hate the term "CD Quality". CD quality is uncompressed 44,100Hz, 16-bit stereo. Anything else is not CD quality...Certainly not mp3. As far as sample rate, there's no benefit to going higher than 44.1kHz for broadcast. Trying to do so with mp3 uses more frequency bandwidth, which means more masking and lower quality.
Emmett
Emmett
For MP3's I use 256k for mono voice and 320k for stereo work. Most all retail audio is done as a WAV file, or AIF for TV. BTW the preferred sample rate for TV production is 48khz AIF files..same size as a WAV (or bigger)
And digital TV audio can get really muddy really fast when using mp3's..not so bad for analog.
And digital TV audio can get really muddy really fast when using mp3's..not so bad for analog.
Doing My Bit
I thought that I'd chime in just to make sure that everybody understands how bit rates are derived.
The rule is that it requires two samples to determine the wavelength (frequency) of a sample. If you sample at 44,100 samples per second, the highest frequency you can determine is 22,050. If you want to accurately represent the amplitude (volume level) of the sample, you'll need more than two samples. Lower frequencies, with longer wavelengths, will be more accurately represented than higher frequencies. Some sampling schemes try to even out this differential by using different scales along the Y-axis to determine amplitude.
The bit rate is determined by multiplying the sampling rate (ex. 44,100 samples/sec for CD quality) x the quantization type (the largest binary number available to describe the amplitude (volume) of an audio sample).
For example, 16-bit audio gives you levels from 0000000000000000 to 1111111111111111 in binary, or 0 - 65534 in decimal. 44,100 sample x 16 bits/sample = 705600 bit/sec., or 705.6 Kbit/sec.
Compression algorithms like MP3 offers varying compression rates. There are also other factors, like how the samples are compressed, and whether compression is at a constant or variable bit rate. The MP3 standard allows quite a bit of freedom with encoding algorithms, and different encoders output different quality, even at similar bit rates.
If you're streaming MP3s at 128K, you're compression ratio is 705.6/128 = 5.125:1. That, of course, doesn't take into consideration the quality of the encoder.
To quote Frauhofer IIS:
In listening tests, MP3 encoders have performed significantly worse than those using more modern compression methods (such as AAC) at low bit rates. In a 2004 public listening test at 32 kbit/s[5], the LAME MP3 encoder scored only 1.79/5 - behind all modern encoders - with Nero Digital HE AAC scoring 3.30/5.
The bottom line is the surfer's motto: There ain't nothin' like a good .wav.
I thought that I'd chime in just to make sure that everybody understands how bit rates are derived.
The rule is that it requires two samples to determine the wavelength (frequency) of a sample. If you sample at 44,100 samples per second, the highest frequency you can determine is 22,050. If you want to accurately represent the amplitude (volume level) of the sample, you'll need more than two samples. Lower frequencies, with longer wavelengths, will be more accurately represented than higher frequencies. Some sampling schemes try to even out this differential by using different scales along the Y-axis to determine amplitude.
The bit rate is determined by multiplying the sampling rate (ex. 44,100 samples/sec for CD quality) x the quantization type (the largest binary number available to describe the amplitude (volume) of an audio sample).
For example, 16-bit audio gives you levels from 0000000000000000 to 1111111111111111 in binary, or 0 - 65534 in decimal. 44,100 sample x 16 bits/sample = 705600 bit/sec., or 705.6 Kbit/sec.
Compression algorithms like MP3 offers varying compression rates. There are also other factors, like how the samples are compressed, and whether compression is at a constant or variable bit rate. The MP3 standard allows quite a bit of freedom with encoding algorithms, and different encoders output different quality, even at similar bit rates.
If you're streaming MP3s at 128K, you're compression ratio is 705.6/128 = 5.125:1. That, of course, doesn't take into consideration the quality of the encoder.
To quote Frauhofer IIS:
In listening tests, MP3 encoders have performed significantly worse than those using more modern compression methods (such as AAC) at low bit rates. In a 2004 public listening test at 32 kbit/s[5], the LAME MP3 encoder scored only 1.79/5 - behind all modern encoders - with Nero Digital HE AAC scoring 3.30/5.
The bottom line is the surfer's motto: There ain't nothin' like a good .wav.
B
BenTehelenbach
Guest
We send out nothing lower than 256kbps. We accept nothing lower than 128kbps. Occasionally, a client or agency will want us to air an mp3 commercial encoded at 40kbps 22050kHz mono...the kind we send them for approval...a sure sign that they're trying to circumvent talent fees.
Re: Doing My Bit
This is, of course, mono, as a stereo wav file is 1411.2kbps, which means greater compression ratios when encoding to mp3. You also must take into account CRC checksums, which help decode the stream without errors, but writing CRC checksums also uses some of the bandwidth. There's the copyright bit, private bit and original bit, which can also be set, and again, uses up some of that space. Other extra information can be stored as well, but everything that is stored to mp3 will degrade the audio quality, as the file size will not change...So something has to give. With wav or aiff, any extra information will be stored separately and will increase file size, but not degrade quality.
The surfers' motto stands! ;D
SirRoxalot said:I thought that I'd chime in just to make sure that everybody understands how bit rates are derived.
The rule is that it requires two samples to determine the wavelength (frequency) of a sample. If you sample at 44,100 samples per second, the highest frequency you can determine is 22,050. If you want to accurately represent the amplitude (volume level) of the sample, you'll need more than two samples. Lower frequencies, with longer wavelengths, will be more accurately represented than higher frequencies. Some sampling schemes try to even out this differential by using different scales along the Y-axis to determine amplitude.
The bit rate is determined by multiplying the sampling rate (ex. 44,100 samples/sec for CD quality) x the quantization type (the largest binary number available to describe the amplitude (volume) of an audio sample).
For example, 16-bit audio gives you levels from 0000000000000000 to 1111111111111111 in binary, or 0 - 65534 in decimal. 44,100 sample x 16 bits/sample = 705600 bit/sec., or 705.6 Kbit/sec.
Compression algorithms like MP3 offers varying compression rates. There are also other factors, like how the samples are compressed, and whether compression is at a constant or variable bit rate. The MP3 standard allows quite a bit of freedom with encoding algorithms, and different encoders output different quality, even at similar bit rates.
If you're streaming MP3s at 128K, you're compression ratio is 705.6/128 = 5.125:1. That, of course, doesn't take into consideration the quality of the encoder.
To quote Frauhofer IIS:
In listening tests, MP3 encoders have performed significantly worse than those using more modern compression methods (such as AAC) at low bit rates. In a 2004 public listening test at 32 kbit/s[5], the LAME MP3 encoder scored only 1.79/5 - behind all modern encoders - with Nero Digital HE AAC scoring 3.30/5.
The bottom line is the surfer's motto: There ain't nothin' like a good .wav.
This is, of course, mono, as a stereo wav file is 1411.2kbps, which means greater compression ratios when encoding to mp3. You also must take into account CRC checksums, which help decode the stream without errors, but writing CRC checksums also uses some of the bandwidth. There's the copyright bit, private bit and original bit, which can also be set, and again, uses up some of that space. Other extra information can be stored as well, but everything that is stored to mp3 will degrade the audio quality, as the file size will not change...So something has to give. With wav or aiff, any extra information will be stored separately and will increase file size, but not degrade quality.
The surfers' motto stands! ;D
No less than 44.1 320k for my standards in MP3 format. I'd go with wave had I the hard drive space at home, but I don't, and as cheap as it is, I don't plan on buying another drive anytime soon (the one I have in here is 140 gigs). I'll just stick with my DVD backups for now.
- Status
- This thread has been closed due to inactivity. You can create a new thread to discuss this topic.