Digital audio meter difference

marvinhello

marvinhello

Well-known member
Hi there I recently noticed something interesting with digital audio meter, when I play the same audio file in Foobar 2000 and Adobe Audition, these two programme give me different dBFS readings, if the audio peaks in Foobar around -20dBFS, then it's around -12dBFS in Adobe Audition, they both have 60dB scale and I've tried 16bit and 24bit file with various sample rate, the difference still exists.

here is another question, in Audition I saw an option of setting playback sample rate, it could be set as high as 192kHz, is there any advantage/disadvantage of setting this to 192? or should I stick to default 44.1kHz? Also in Windows playback devices properties, there is a setting about sample rate in shared mode, do I set this to highest 24/192 as well? thanks:cheesy:
 
marvinhello said:
Hi there I recently noticed something interesting with digital audio meter, when I play the same audio file in Foobar 2000 and Adobe Audition, these two programme give me different dBFS readings, if the audio peaks in Foobar around -20dBFS, then it's around -12dBFS in Adobe Audition, they both have 60dB scale and I've tried 16bit and 24bit file with various sample rate, the difference still exists.

here is another question, in Audition I saw an option of setting playback sample rate, it could be set as high as 192kHz, is there any advantage/disadvantage of setting this to 192? or should I stick to default 44.1kHz? Also in Windows playback devices properties, there is a setting about sample rate in shared mode, do I set this to highest 24/192 as well? thanks:cheesy:
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The standard sample rate for audio for video or film is 48kHz, 16bit. 44.1 is the standard for music CDs. Use 48. Some workers use 96kHz for critical music and effects master recordings, converting to 48 when subsequently incorporating it into a video project. Anything beyond 96 is pointless and unless you have really top-shelf professional gear throughout your recording chain even 96 is overkill.

Don't know anything about Foobar or its metering system but looking at the website it appears that the meters are VU averaging meters. Adobe Audition, OTOH, uses peak-reading meters. What you're observing is due to the differing response characteristics of the two meter types. Averaging meters integrate the signal level over a period of time, typically 300 milliseconds, and show the RMS (Root Mean Square) average value of the waveform voltage over that time period. A peak-reading tracks the approximate instantaneous value of the signal voltage, typically over about a 10 millisecond rise time. If you send a pure sine wave tone to a pair of meters, one averaging and the other peak-reading, they will read the same. But for real world complex signals this isn't true. With a complex waveform such as real-world sounds, speech, or pink noise (which simulates the spectral distribution of real-world signals), the RMS average signal voltage is approxmately 0.7 times the voltage of the waveform peaks. So if you send speech or a pink noise signal to that same pair of meters, when the averaging meter is reading -20dBFS the peak meter will read about -12dBFS, the voltage represented by -20 being about 0.7 times the voltage represented by -12. Note, because the peak meter can't be truly instantaneous or they'd move so fast you'd never be able to read them, the ACTUAL transient peaks of the waveform can be 4 to 6 dB higher than even the peak meter can display. Most cameras and digital devices use peak meters, professional equipment such as the Sound Devices mixers will often have user-selectable meter ballistics.

Take a look at the videos on this site for an interesting demonstration ... http://dorrough.com/video.html
 
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Peak meters also typically have a slight hold time on peak readings 100-300 milliseconds, so the level can be read by the operator. In some systems the hold time is variable too. If a higher peak occurs during hold, the meter reading adjusts to reflect the new peak instantly and hold starts over.
 
Razz16mm said:
Peak meters also typically have a slight hold time on peak readings 100-300 milliseconds, so the level can be read by the operator. In some systems the hold time is variable too. If a higher peak occurs during hold, the meter reading adjusts to reflect the new peak instantly and hold starts over.
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Yep, true, but be careful. Even with a peak-hold meter, rapid transients on the order of a few cycles duration are still too fast to register. One of the reasons I worry when I read posts claiming they record with the meter's tickling -3 or some such - those rapid transients 4 to 6 dB over the meter reading that aren't registering can still produce audible clipping if they hit 0dBFS.
 
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The only time I've used 192Khz is for recording certain sound effects I know I'm going to do some heavy duty time expansion on. Some people record all their sound effects at 192Khz these days. At normal speed it's hard, if not impossible, to tell the difference between 96Khz and 192Khz, but once you start to expand the audio file, you'll really tell a difference. If you aren't planning on doing some crazy sound design, then 192Khz is not necessary. It just takes up more disk space.
 
davefisk said:
The only time I've used 192Khz is for recording certain sound effects I know I'm going to do some heavy duty time expansion on. Some people record all their sound effects at 192Khz these days. At normal speed it's hard, if not impossible, to tell the difference between 96Khz and 192Khz, but once you start to expand the audio file, you'll really tell a difference. If you aren't planning on doing some crazy sound design, then 192Khz is not necessary. It just takes up more disk space.
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Yep, plus a lot of software/plugins still don't support 192 kHz. 96 kHz is more than generous and to me its main purpose is usage with plugins that don't oversample. But this is 2011, most plugins/software digital signal processors do oversample. I still record at 24-bit/96 kHz anyway though because you never know when a producer wants the actor to be auto-tuned in the scene! ;-)
 
adamr316 said:
Yep, plus a lot of software/plugins still don't support 192 kHz. 96 kHz is more than generous and to me its main purpose is usage with plugins that don't oversample. But this is 2011, most plugins/software digital signal processors do oversample. I still record at 24-bit/96 kHz anyway though because you never know when a producer wants the actor to be auto-tuned in the scene! ;-)
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How about playback sample rate? in Adobe Audition, when setting "Audio Hardware", there is an option of setting playback sample rate, from 44.1 all the way up to 192, should I adjust this according to specific project or just leave it on highest setting? is the concept of "bit-perfect" really important?
 
marvinhello said:
How about playback sample rate? in Adobe Audition, when setting "Audio Hardware", there is an option of setting playback sample rate, from 44.1 all the way up to 192, should I adjust this according to specific project or just leave it on highest setting? is the concept of "bit-perfect" really important?
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You should use the playback sample rate that will be used in the final output file for your project settings in Audition. For video production, that is ALWAYS 48kHz. The above discussions about using 96 or higher refer to making original recordings, not the timeline in your audio workstation or video NLE. If you're outputting to an audio CD the project sample rate should be 44.1, if you're outputting audio for video, whether broadcast or DVD, the sample rate should be 48kHz. Recordings made at other sample rates need to be converted to the timeline rate when brought into the timeline.
 
Steve House said:
You should use the playback sample rate that will be used in the final output file for your project settings in Audition. For video production, that is ALWAYS 48kHz. The above discussions about using 96 or higher refer to making original recordings, not the timeline in your audio workstation or video NLE. If you're outputting to an audio CD the project sample rate should be 44.1, if you're outputting audio for video, whether broadcast or DVD, the sample rate should be 48kHz. Recordings made at other sample rates need to be converted to the timeline rate when brought into the timeline.
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Steve, in what circumstances then would you record a higher sample rate than 48kHz, since you are converting to 48kHz anyway for video? I'm just trying to get my head around when 96kHz or higher would be practical for original recording sample rate.
 
tirapell said:
Steve, in what circumstances then would you record a higher sample rate than 48kHz, since you are converting to 48kHz anyway for video? I'm just trying to get my head around when 96kHz or higher would be practical for original recording sample rate.
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I might use a higher sample rate if I anticpated I would be mixing a large number of digital sources or applying a large amount of digital processing prior to incorporating the track into the final timeline. For typical dialog and music recording 48kHz is just fine. The sample rate's immediate effect is on the maximum frequncy that can accurately be recorded, the so-called Nyquist Limit which says that the highest frequency that can be faithfully recorded is equal to 1/2 the sample rate. 1/2 of 48k is 24,000Hz, well above the s0-to-20kHz range limits of human hearing. Some audiophiles insist that the presence of high order harmonics above the audible range still affects the timbre of musical instruments so you'll most often find high sample-rates used for audiophile music recordings. How much better they are than conventional recordings remains to be seen.
 
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I'll jump in. I am not a big high sample rate fan BUT for some FX it works well. One of the most common manipulations in sound post is slowing things down. In analog days that worked well but digital can get "grainy' real fast. One way around that is to use a high sample rate. So if I know I'm going to slow something down I will record at a higher rate because the slow down will be smoother.

So it lives in my library at 96 but it gets slowed down and used at 48.
 
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