Taperssection.com
Gear / Technical Help => Recording Gear => Topic started by: gaijin on January 19, 2022, 08:05:48 PM
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https://www.newsshooter.com/2022/01/19/zoom-f3/
Curious about the groups thoughts on this one. I have not been a huge fan of the preamps on the Zooms historically. Don't love the lack of a mini jack line in either. Curious if anyone has an opinion on this vs a MixPre3ii, the new Tascam Portacapture, or any similar devices.
Portable Recorder
Form Factor Handheld Recorder
Primary Use Applications Audio-for-Video
Number of Tracks 2
Number of Input Channels 2
Max Sample Rate/Resolution 192 kHz / 32-Bit
Microphone None
Number of Mic Preamps 2
Built-In Speaker No
Memory Card Support microSDHC (4 GB to 32 GB)
microSDXC (64 GB to 1 TB)
Internal Storage None
Special Features Remote Control Capable (Remote Available Separately)
Recording
Audio File Formats BWF, iXML
Sample Rates 44.1 / 48 / 88.2 / 96 / 192 kHz
Bit Depth 32-Bit
Signal Processing & FX None
Timecode Support Yes
Included Software None
Connectivity
Analog I/O 2 x XLR 3-Pin Female Balanced Line/Mic Input (Lockable)
1 x 1/8" / 3.5 mm TRS Female Unbalanced Line Output
1 x 1/8" / 3.5 mm TRS Female Unbalanced Headphone Output
Digital I/O None
Phantom Power +24 / 48 V, Selectable On/Off
Phantom Power Current 10 mA per Channel
Host Connection 1 x USB Type-C (USB 2.0)
Audio Interface Capability Stereo Only
Performance
Max Input Level Mic Inputs:
+4 dBu
Line Inputs:
+24 dBu
Max Output Level Line Outputs:
+1 dBu
Headphone Output Power 50 mW per Channel into 32 Ohms
Impedance Mic/Line Inputs:
≥3 Kilohms
Line Outputs:
200 Ohms
Headphone Outputs:
200 Ohms
EIN ≤-127 dBu
Power
Power Options Battery, USB Bus Power, or AC/DC Power Adapter
Battery Type 2 x AA (Included)
Approximate Battery Life 48 kHz/32-Bit:
8 Hours (AA Alkaline, 2 Channels, Recording)
8.5 Hours (AA NiMH, 2 Channels, Recording)
18 Hours (AA Lithium, 2 Channels, Recording)
AC/DC Power Adapter 5 VDC at 1 A (Not Included)
Physical
Included Accessories Batteries
Mounting Options None
Dimensions 3 x 1.9 x 3" / 75 x 47.8 x 77.3 mm
Weight 8.5 oz / 242 g (with Batteries)
Available End of Feb 2022
MSRP $349.99
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That thing looks pretty sweet. I would be afraid of losing it though. If it sounds really good and is affordable, I would up fr trying it. I wonder if it can do 8 hours with internal batteries on P48?
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does usb audio. would be a nice small recorder to pair with my sonosax m2d2 vs phone
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This looks really nice.
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So tiny. I was discussing with kindms that the apparent market for this is amateur video and field reporter types who work with small crews. Although it fits on a boom pole, it is certainly not a device for large scale movie/film production as the timecode isn't universal.
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Japanese site is charging ~$310 for it, though it might be different here:
https://www.kizaiya.jp/shopdetail/000000001693/ (https://www.kizaiya.jp/shopdetail/000000001693/)
Other than 32 bit, not sure why you would run it with an M2D2 since the A/D on that has to be better than the Zoom's, even an iPod Touch makes a better bitbox. But 32 bits helps if you're trying to record crickets near a nuclear explosion, I guess.
Jeff
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B&H now has it listed for end of February, no price yet.
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Other than 32 bit, not sure why you would run it with an M2D2 since the A/D on that has to be better than the Zoom's, even an iPod Touch makes a better bitbox. But 32 bits helps if you're trying to record crickets near a nuclear explosion, I guess.
Jeff
We were thinking using USB in does not use the AD on the f3. Probably incorrect, but maybe a menu setting?
Currently, kindms is using an old, no telephone line iPhone7S. Seemed to work GREAT the first time we used it.
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I don't think it's as small as it seems. Physical dimensions listed as 3" x 3" x 2". By comparison, the F6 is 4" x 4.7" x 2.5". So with an inch less of width and 1.7" less of length, you get four fewer preamps. I know that when stealthing every inch counts but I don't think this is tiny. It won't slip into your pocket, although it will mount on your belt
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Japanese site is charging ~$310 for it, though it might be different here:
https://www.kizaiya.jp/shopdetail/000000001693/ (https://www.kizaiya.jp/shopdetail/000000001693/)
Other than 32 bit, not sure why you would run it with an M2D2 since the A/D on that has to be better than the Zoom's, even an iPod Touch makes a better bitbox. But 32 bits helps if you're trying to record crickets near a nuclear explosion, I guess.
Jeff
To use as a dedicated USB audio recorder. Phone works fine but I like the idea of a dedicated recorder. so A/D in the m2d2 and just sending 1s and 0s to the zoom over USB.
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Got a love hate relationship with Zoom gear.
Love the small size and also the locking connectors on the F1 (stereo) and F2 (float) rigs.However, I've had the USB fail on a Zoom F1 and an H3-VR already.
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I have not been a huge fan of the preamps on the Zooms historically.
https://taperssection.com/index.php?topic=186654.0
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Other than 32 bit, not sure why you would run it with an M2D2 since the A/D on that has to be better than the Zoom's, even an iPod Touch makes a better bitbox. But 32 bits helps if you're trying to record crickets near a nuclear explosion, I guess.
Jeff
Comparing specs, the F3 and F6 look like they use same preamp circuit, but obviously different ADC.
I can believe that Sonosax has better A/D than Zoom, but I don't know how the iPod Touch would be a better bitbox than a dedicated audio recorder. Or are you just saying that in the context of using the M2D2 in front of it?
Regarding your "crickets" comment: Don't you own an F6? Mine has made what I do so much easier, specifically because of the auto-ranging dual DACs and 32 FP setup. When you're often recording your own performances, and/or don't have a rehearsal to set levels, you can just hit record and go without concern for limiters, safety tracks, etc. I feel like that this has been discussed to death on various threads, with most of the criticism coming from people who are making assumptions about these units without any firsthand experience.
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I have not been a huge fan of the preamps on the Zooms historically.
https://taperssection.com/index.php?topic=186654.0
The Zoom H series preamps are quite noisy, and deserving of criticism. F6 and F8 in a different league, and are now on par with Sound Devices and other pro units in all areas except max input level. I haven't heard the F1 / F2 so I can't speak to those. As I said earlier, the F3 looks like it might use the same preamps as the F6.
If you need some proof: WiFiJeff posted some excellent solo piano recordings a while back made with an F6 (which I can't locate; maybe he will be willing to repost).
Here are some choir recordings of mine, all with an F6. (https://www.dropbox.com/sh/auvka6syq8jgopr/AAArGSLuwKHxTmim80n6vFIDa?dl=0)
I also did some measurements of my F6 before/after upgrading firmware (https://taperssection.com/index.php?topic=190161.msg2358002#msg2358002). There was a low-frequency noise issue in the F6 that you will see has been corrected by the latest firmware (even though the issue was totally inaudible).
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One concerning thing I just caught in the specs:
Phantom Power
Channel total 10 mA or less
Typically you see a spec of of 10 mA max for each channel. Something to be aware of when selecting mics. Most good mics don't have very high current consumption anymore, but you couldn't use a pair of AKG C414 or Gefell M221, for example.
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The Zoom H series preamps are quite noisy, and deserving of criticism. F6 and F8 in a different league, and are now on par with Sound Devices and other pro units in all areas except max input level.
Good point, I overlooked that distinction.
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I can believe that Sonosax has better A/D than Zoom, but I don't know how the iPod Touch would be a better bitbox than a dedicated audio recorder. Or are you just saying that in the context of using the M2D2 in front of it?
Just in the context of running the M2D2. If you run the M2D2 digital-out, you might as well run to the iPod touch or an iPhone as the M2D2 won't do 32 bits. If you run analog-out from the M2D2 you would get the A/D of the F3. My iPhone XR running Metarecorder runs for about 10-11 hours, when powering a DPA MMA-A and 4060 mics, and I calculate it would run around 13-14 hours with the M2D2. An iPod Touch 7 recording (with Metarecorder) from an M2D2 (the M2D2 battery powering the mics and its own amp/AD) runs about 5 hours, which is a bit longer than the M2D2 gets on its internal battery.
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does usb audio. would be a nice small recorder to pair with my sonosax m2d2 vs phone
I doubt very much whether it records from its USB interface. My reading of the available info is that it will be just the same as any other Zoom recorder, namely that you can use it as an interface to a computer, allowing the F3 mic inputs to be recorded on the PC, and playback on the PC could be heard on the Zoom. It won't have a digital input for recording.
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I found the manual online - concerning the USB interface, it says (according to Google Translate) -
"Connect to a PC or smartphone / tablet to connect with an SD card reader and audio interface
Can be used. It supports USB bus power operation."
In the menus you can select whether you want to connect to PC/Mac, or to smartphone / tablet.
There are diagrams showing connection to PC/Mac or smartphones/tablets but no reference to connecting a USB-C audio device.
"You can set the loopback and monitor sound when using F3 as an audio interface.
Set loopback
Mix the playback sound of your computer or smartphone / tablet with the input sound of F3, and then use your computer or smartphone again.
You can send it to a phone / tablet (loopback).
You can add narration to the music played on your computer, record it on your computer, or stream it. "
Link to the manual - https://zoomcorp.com/media/documents/J_F3.pdf
Of course it might be possible to turn it into a USB-C audio recorder by FW update, but typically that would be to enable a USB mic to be connected. I don't think that would be something that most users of such a device would want to do, so it probably wouldn't be worth their effort.
I'm aware of only one device on the market that records from USB-C audio devices (obviously apart from PC/Mac/Android/iOS devices) and that's the DJI Action 2 video camera. However that device only records to AAC format with no options to change audio settings, and while the audio quality is fine in relation to the use case and design intent of the device, it has no particular advantage over phones for location recording of audio.
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I can believe that Sonosax has better A/D than Zoom, but I don't know how the iPod Touch would be a better bitbox than a dedicated audio recorder. Or are you just saying that in the context of using the M2D2 in front of it?
Just in the context of running the M2D2. If you run the M2D2 digital-out, you might as well run to the iPod touch or an iPhone as the M2D2 won't do 32 bits. If you run analog-out from the M2D2 you would get the A/D of the F3. My iPhone XR running Metarecorder runs for about 10-11 hours, when powering a DPA MMA-A and 4060 mics, and I calculate it would run around 13-14 hours with the M2D2. An iPod Touch 7 recording (with Metarecorder) from an M2D2 (the M2D2 battery powering the mics and its own amp/AD) runs about 5 hours, which is a bit longer than the M2D2 gets on its internal battery.
I know my AD and how to maximize it. I again was simply liking the idea of a small dedicated recorder that i can use with the USB out on the M2D2. I have a free iphone7 that i have used successfully and its great. I can run it all day long etc as you say. But at the end of the day its also a phone and it cannot accept anything but USB audio. A small dedicated recorder that can do both USB audio as well as have other features I might find useful in small recorder is of value. Plus as well as the iphone works etc, I find the adapter and cabling a little clunky. So reducing that would be cool. I dont know if it would be worth $300+ cool. But a small recorder that has a USB and/or Digital in would be awesome for me. I have considered building a pi to do it but it would be bigger and more costly than a phone etc
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I found the manual online - concerning the USB interface, it says (according to Google Translate) -
"Connect to a PC or smartphone / tablet to connect with an SD card reader and audio interface
Can be used. It supports USB bus power operation."
In the menus you can select whether you want to connect to PC/Mac, or to smartphone / tablet.
There are diagrams showing connection to PC/Mac or smartphones/tablets but no reference to connecting a USB-C audio device.
"You can set the loopback and monitor sound when using F3 as an audio interface.
Set loopback
Mix the playback sound of your computer or smartphone / tablet with the input sound of F3, and then use your computer or smartphone again.
You can send it to a phone / tablet (loopback).
You can add narration to the music played on your computer, record it on your computer, or stream it. "
Link to the manual - https://zoomcorp.com/media/documents/J_F3.pdf
Of course it might be possible to turn it into a USB-C audio recorder by FW update, but typically that would be to enable a USB mic to be connected. I don't think that would be something that most users of such a device would want to do, so it probably wouldn't be worth their effort.
I'm aware of only one device on the market that records from USB-C audio devices (obviously apart from PC/Mac/Android/iOS devices) and that's the DJI Action 2 video camera. However that device only records to AAC format with no options to change audio settings, and while the audio quality is fine in relation to the use case and design intent of the device, it has no particular advantage over phones for location recording of audio.
the sound devices mix pre II series do it
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I found the manual online - concerning the USB interface, it says (according to Google Translate) -
"Connect to a PC or smartphone / tablet to connect with an SD card reader and audio interface
Can be used. It supports USB bus power operation."
In the menus you can select whether you want to connect to PC/Mac, or to smartphone / tablet.
There are diagrams showing connection to PC/Mac or smartphones/tablets but no reference to connecting a USB-C audio device.
"You can set the loopback and monitor sound when using F3 as an audio interface.
Set loopback
Mix the playback sound of your computer or smartphone / tablet with the input sound of F3, and then use your computer or smartphone again.
You can send it to a phone / tablet (loopback).
You can add narration to the music played on your computer, record it on your computer, or stream it. "
Link to the manual - https://zoomcorp.com/media/documents/J_F3.pdf
Of course it might be possible to turn it into a USB-C audio recorder by FW update, but typically that would be to enable a USB mic to be connected. I don't think that would be something that most users of such a device would want to do, so it probably wouldn't be worth their effort.
I'm aware of only one device on the market that records from USB-C audio devices (obviously apart from PC/Mac/Android/iOS devices) and that's the DJI Action 2 video camera. However that device only records to AAC format with no options to change audio settings, and while the audio quality is fine in relation to the use case and design intent of the device, it has no particular advantage over phones for location recording of audio.
the sound devices mix pre II series do it
I'm really curious about the details of this. Does this approach work with any proper USB input device? For example, if I connect a Shure MV-88+ to the MixPre3,6-ii, will I be able to record a set of tracks using it's USB connection? All the references I have seen to the USB audio connectivity is in the context of to PC connections. Is there somewhere that explains how the functionality you're describing works?
Thanks in advance for any detail you can share.
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I found the manual online - concerning the USB interface, it says (according to Google Translate) -
"Connect to a PC or smartphone / tablet to connect with an SD card reader and audio interface
Can be used. It supports USB bus power operation."
In the menus you can select whether you want to connect to PC/Mac, or to smartphone / tablet.
There are diagrams showing connection to PC/Mac or smartphones/tablets but no reference to connecting a USB-C audio device.
"You can set the loopback and monitor sound when using F3 as an audio interface.
Set loopback
Mix the playback sound of your computer or smartphone / tablet with the input sound of F3, and then use your computer or smartphone again.
You can send it to a phone / tablet (loopback).
You can add narration to the music played on your computer, record it on your computer, or stream it. "
Link to the manual - https://zoomcorp.com/media/documents/J_F3.pdf
Of course it might be possible to turn it into a USB-C audio recorder by FW update, but typically that would be to enable a USB mic to be connected. I don't think that would be something that most users of such a device would want to do, so it probably wouldn't be worth their effort.
I'm aware of only one device on the market that records from USB-C audio devices (obviously apart from PC/Mac/Android/iOS devices) and that's the DJI Action 2 video camera. However that device only records to AAC format with no options to change audio settings, and while the audio quality is fine in relation to the use case and design intent of the device, it has no particular advantage over phones for location recording of audio.
the sound devices mix pre II series do it
I'm really curious about the details of this. Does this approach work with any proper USB input device? For example, if I connect a Shure MV-88+ to the MixPre3,6-ii, will I be able to record a set of tracks using it's USB connection? All the references I have seen to the USB audio connectivity is in the context of to PC connections. Is there somewhere that explains how the functionality you're describing works?
Thanks in advance for any detail you can share.
It would seem I am WRONG. I was under the impression that they do allow USB audio in but from what Im reading in the manual it would seem that isn't correct. It seems it does require a computer as others have mentioned up thread about other devices.
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I'd lay money that before long, recording from USB devices will be implemented on location recording equipment - persuasive use case would be recording from kit like the Rode Wireless Go 2 USB output, or from the equivalent DJI device. Or from the Sennheiser Ambeo binaural headset! If DJI can do this with their Action 2 camera, then there's clearly no basic technical problem - unless there would be a conflict with using the recorder as an audio interface to a PC. Maybe you can't do both in one device.
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Almost here and MSRP
https://www.bhphotovideo.com/c/product/1686005-REG/zoom_zf3_f3_digital_field_recorder.html
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Almost here and MSRP
https://www.bhphotovideo.com/c/product/1686005-REG/zoom_zf3_f3_digital_field_recorder.html
Nice! Added details to the first post.
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What's up with that screen, though? It's 2022 and all. It would probably only add a couple of bucks to the list price to have something a little more contemporary. It's worse than a first generation GameBoy...
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One concerning thing I just caught in the specs:
Phantom Power
Channel total 10 mA or less
Typically you see a spec of of 10 mA max for each channel. Something to be aware of when selecting mics. Most good mics don't have very high current consumption anymore, but you couldn't use a pair of AKG C414 or Gefell M221, for example.
Looks like I was way off base on this. It's 10 mA per channel, as you would expect.
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I have asked B&H whether there is any support for M/S mics - given that it's designed partly for video I would be surprised if it didn't even decode M/S to X/Y for headphone monitoring. I really hope they have not overlooked this - it could be a dealbreaker for me (which would save me some money of course!).
[Edit - reading the English manual at https://zoomcorp.com/media/documents/E_F3.pdf it looks like there is no M/S support. That astonishes me.]
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I have been out of the loop for some years regarding 32bit float.
Is the format now widely adapted, also in the entry level DAW's?
When the MixPre II series and F6 came out, I recall I checked and found out I couldn't work on the files in GarageBand or Hindenburg Lite.
Has that changed?
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I have been out of the loop for some years regarding 32bit float.
Is the format now widely adapted, also in the entry level DAW's?
When the MixPre II series and F6 came out, I recall I checked and found out I couldn't work on the files in GarageBand or Hindenburg Lite.
Has that changed?
I can't speak to those particular DAWs, but Reaper has handled float-point files without issues for years.
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I have asked B&H whether there is any support for M/S mics - given that it's designed partly for video I would be surprised if it didn't even decode M/S to X/Y for headphone monitoring. I really hope they have not overlooked this - it could be a dealbreaker for me (which would save me some money of course!).
[Edit - reading the English manual at https://zoomcorp.com/media/documents/E_F3.pdf it looks like there is no M/S support. That astonishes me.]
I have never liked the idea of applying a M/S decoding matrix straight to the recorder, as it defeats one of the most major benefits of M/S recording - being able to change the imaging in post. I completely agree, however, that having a decoding matrix for monitoring is a necessary feature, especially for boom pole operators which this unit seems perfectly designed for. If you are recording 32-bit float, there is no need to monitoring levels, but it would be nice to at least hear what's going into the mics in a "default" 50/50 M/S decode.
This is something that could probably be added via a firmware update.
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I have been out of the loop for some years regarding 32bit float.
Is the format now widely adapted, also in the entry level DAW's?
When the MixPre II series and F6 came out, I recall I checked and found out I couldn't work on the files in GarageBand or Hindenburg Lite.
Has that changed?
I can't speak to those particular DAWs, but Reaper has handled float-point files without issues for years.
Thanks. I think that was the case at the time, and I checked it out but found the UI unintuitive.
I'll look for a 32bit float file and see how it imports - not much info in he respective applications help fuction.
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I have asked B&H whether there is any support for M/S mics - given that it's designed partly for video I would be surprised if it didn't even decode M/S to X/Y for headphone monitoring. I really hope they have not overlooked this - it could be a dealbreaker for me (which would save me some money of course!).
[Edit - reading the English manual at https://zoomcorp.com/media/documents/E_F3.pdf it looks like there is no M/S support. That astonishes me.]
I have never liked the idea of applying a M/S decoding matrix straight to the recorder, as it defeats one of the most major benefits of M/S recording - being able to change the imaging in post.... [snipped]
I am very sure that we are on the same page in that respect - but I will none the less mention that there are (for me at least) two merits in MS recordings. One is that they can be decoded to provide whatever width is appropriate. This can vary from mono (for speech), through various degrees of stereo for music or ambience, or you can have the side mics only and have an interviewer and victim either side of the fig of 8 capsule. But there's actually no reason why normal X/Y recordings shouldn't be processed for width in post-production much as an MS recording is processed - you just need a plugin like Voxengo Msed to convert from XY to MS, vary the width, then reconvert back to XY. However, the difference between an MS pair and an XY pair is that the MS pair has that front facing mic for most appropriate mono recording, and the fig of 8 mic for, well, fig of 8 applications.
So - when using an MS mic with a recorder which will decode the signal to XY before you record, there's no theoretical difference between that approach and the alternative of recording the MS pair 'as is' undecoded. Pass either signal through MSED and you will end up with the same degree of choice over width, albeit by first converting the already dematrixed XY version back to MS, tweaking it, and converting to XY again, as compared with using the native MS recording 'as is' and tweaking it for width when dematrixing it with the plugin.
But having said all that, indeed the purist approach is simply to monitor in XY but record in MS, as you say.
Show me a nit and I will pick it...
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Ozpeter is right. We can encode any stereo track left/right to mid/side and back to left/right without loss. But when we do editing in mid/side, it's good that the stereo doesn't have a time difference - it is recorded in coincident config (xy). If the stereo has a time differences - spaced config (din, ortf, ab ...) some degradation will occur when editing in mid/side and converting back to left/right. But it's not always disturbing, sometimes I do mid/side editing for spaced configs.
I think stereo mp3 is also encoded in mid/side and players decode it internally to left/right. When I created a stereo mp3 that contained only a mono recording but in two tracks, the file was still half the size. I assume there is zero side channel and only mid channel is compressed.
Recording with fig8+mid or with directional microphones in xy works on the same principle. We can get the same results with a pair of directional mics in xy as with fig8+mid. Very nice explanation is in this thread. At the end, our friends are explaining what are the practical differences when recording with mid and fig.8 vs directional mics in xy.
https://taperssection.com/index.php?topic=193204.msg2320506#msg2320506
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This looks like the first proper review:
https://www.audiotechnology.com/reviews/zoom-f3
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What's up with that screen, though? It's 2022 and all. It would probably only add a couple of bucks to the list price to have something a little more contemporary. It's worse than a first generation GameBoy...
I guess they thought that you'll get all the resolution and color you want on the smartphone you use to control it with.
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^ I suppose, but, if I am standing right next to the thing, I would prefer not to deal with possible latency or connection issues and just monitor/control my recording using the built-in features. I doubt a more modern screen would have added much cost.
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And think of all those beautiful, maybe screen-related, ultra-sonic hum resonances you'd be able to get, just like the Tascam X8!
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This looks like the first proper review:
https://www.audiotechnology.com/reviews/zoom-f3
I wish the reviewer had told people the whole story: It's 32-bit floating point, being fed by multiple auto-ranging ADCs. It's explained pretty well in the manual. He also says "dynamic range will be limited by your microphone and not your recorder." Not the whole story either - if you have a sensitive mic with a hot output recording a loud source, you can overload the preamp ahead of the ADC. (I had to be reminded of this in the F6 thread.)
Still, after using my F6 exclusively in 32-bit FP mode for over 2 years, I am happy to see Zoom drop the level controls for this simple 2-channel recorder. They are completely unnecessary for recording with this kind of setup, but hopefully there is plenty of adjustment for monitoring for those who need to do so.
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This unit looks very interesting, I love small and simple. I will likely end up grabbing one to play with. I'm confused as to whether or not you can actually even (although you don't 'need' to) adjust the recording signal level? The full operation manual states:
Recording
The F3 records in 32-bit float format, so the input gain does not need to be adjusted. Depending on the input signals, however, waveforms shown on the display could appear too small or too large, making them difficult to check.
By changing the waveform display magnification, input waveforms can be adjusted to sizes that are easy to check. The magnification rate can be set separately for each input. The volume will also be changed according to the waveform magnification rate.
NOTE:
• The magnification rate can be set in 11 steps: ×1 , ×2, ×4, ×8, ×16, ×32, ×64, ×128, ×256, ×512 and ×1024. • Be careful with the volume if you are monitoring the input sound with headphones, for example.
• Changing the magnification rate, even in the middle of recording, will not affect the recording level.
But the quick tour manual says:
Adjusting waveform amplification
• Changing the amplification will change the volume and be applied to the recording data.
• Since the F3 records using 32-bit float WAV format, if monitored audio sounds distorted, lowering the volume of the recording file can restore the waveform to a state without clipping.
Seems conflicting?
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This unit looks very interesting, I love small and simple. I will likely end up grabbing one to play with. I'm confused as to whether or not you can actually even (although you don't 'need' to) adjust the recording signal level? The full operation manual states:
Recording
The F3 records in 32-bit float format, so the input gain does not need to be adjusted. Depending on the input signals, however, waveforms shown on the display could appear too small or too large, making them difficult to check.
By changing the waveform display magnification, input waveforms can be adjusted to sizes that are easy to check. The magnification rate can be set separately for each input. The volume will also be changed according to the waveform magnification rate.
NOTE:
• The magnification rate can be set in 11 steps: ×1 , ×2, ×4, ×8, ×16, ×32, ×64, ×128, ×256, ×512 and ×1024. • Be careful with the volume if you are monitoring the input sound with headphones, for example.
• Changing the magnification rate, even in the middle of recording, will not affect the recording level.
But the quick tour manual says:
Adjusting waveform amplification
• Changing the amplification will change the volume and be applied to the recording data.
• Since the F3 records using 32-bit float WAV format, if monitored audio sounds distorted, lowering the volume of the recording file can restore the waveform to a state without clipping.
Seems conflicting?
You can change the volume or the outputted signal, but not the gain which is the actual recorded level. That is why the wave is shown versus bars. I like that idea. That said if you go with a much bigger or smaller wave form that optimal for the end product, you can easily adjust in post and the sound will be just fine. (And yes I understand many have written about exceptions and possible problems where there could be distortion for one reason or another but in my few years of using the Zoom F6 and MixPre 6II for all kinds of music, blasting loud and painfully soft, I have yet to experience anything less than a stellar sounding recording.)
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OK, so it works like the F6 in the sense that you cannot adjust preamp gain, but you can adjust the post-ADC fader level (REC level as the F6 docs call it).
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Thanks I think that helps clarify. So changing the magnification level indeed affects the recorded waveform level? I'm assuming that with amplified PA music low levels won't be an issue, but in the case of recording softer material is it better to try to get the volume from the machine while recording (via magnification), or boost in post in terms of noise/artifacts? I'm guessing since it's digital amplification there wouldn't be much difference?
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Recording will be at a fixed level. Monitoring and playback will follow whatever amplification you apply to the waveform display. "• Changing the amplification will change the volume and be applied to the recording data." means that the metadata for the recording will include the current amplification so it will play back (in the device at least) at that volume if you close the file and reopen it. I don't know whether DAWs will also pick up that volume data. I suspect there's a difference in meaning between recordED data - meaning the sound recording - and recordING data - meaning the metadata.
I don'r actually know any more than anyone else whether my interpretation of the manual is correct but if you bear in mind that "level" means input and "volume" means playback, then I'm pretty sure I'm right.
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Recording will be at a fixed level. Monitoring and playback will follow whatever amplification you apply to the waveform display. "• Changing the amplification will change the volume and be applied to the recording data." means that the metadata for the recording will include the current amplification so it will play back (in the device at least) at that volume if you close the file and reopen it. I don't know whether DAWs will also pick up that volume data. I suspect there's a difference in meaning between recordED data - meaning the sound recording - and recordING data - meaning the metadata.
I don'r actually know any more than anyone else whether my interpretation of the manual is correct but if you bear in mind that "level" means input and "volume" means playback, then I'm pretty sure I'm right.
That's an interesting theory, but the WAV file header does not have a place to store playback volume info. The F3 would have to create separate metadata files to do what you describe.
https://docs.fileformat.com/audio/wav/ (https://docs.fileformat.com/audio/wav/)
The block diagram on pg. 102 of the manual gives a little more clarity, IMO. Notice that Waveform Amplification (shown as a level trim) is directly in the path of everything that comes after it; not just Waveform Display which is simply a tap off of the main path. To me, this makes it clear that adjusting that setting does in fact affect the recorded level, in the same way you can with the F6 in 32FP when setting the trim pot controls to REC LEVEL. See the attached screenshots of the F3 and F6 block diagrams. F3 Waveform Display and F6 REC Level are in the same place in their respective signal chains. While this isn't hard proof, I think it makes sense that Zoom would carry over this implementation from the F6 to the F3.
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Well - it looks like I'll just have to buy one to find out - seems as good an excuse as any to me... :) I note that the recorder uses iXML files which, having read about them for 30 seconds, seem potentially to cater for storing the waveform amplificaton data. And it seems counterproductive that if you wanted a closer look at the waveform during recording, it would affect the recording level and thus screw up the recording. And the manual says it doesn't. So - colour me confused.... However, given that you have an F6 and I only have an F1, you know more about 32 bit recording than I do!
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And it seems counterproductive that if you wanted a closer look at the waveform during recording, it would affect the recording level and thus screw up the recording.
Totally. And if that is the case hopefully it will save last magnification settings used so you don’t have to fiddle too much with re-setting each channel if you repeatedly record the same general volume of material, as the benefit of this deck for me would be to ‘set on the high side and forget it’ unless I was recording much softer material. I have an email into them about that.
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Gain is fixed, so it will not matter if you oversize of undersize the wave file during recording. That is only showing you the output file. You just adjust in post and all will be fine.
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Well - it looks like I'll just have to buy one to find out - seems as good an excuse as any to me... :) I note that the recorder uses iXML files which, having read about them for 30 seconds, seem potentially to cater for storing the waveform amplificaton data. And it seems counterproductive that if you wanted a closer look at the waveform during recording, it would affect the recording level and thus screw up the recording. And the manual says it doesn't. So - colour me confused.... However, given that you have an F6 and I only have an F1, you know more about 32 bit recording than I do!
Not to belabor the point, but as cd2go pointed out Zoom actually say that it does and also that it does not affect the recorded level, depending on if you look in the Quick Start or the full manual. It's possible we're all wrong!
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Without reading again what the manual(s) said, I think it's less confusing if you treat "recording" and "recorded" as different things, and "level" and "volume" as different things also - but indeed, the verdict will have to wait until a competent reviewer gets their hands on one.
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Let’s really beat this thing to death, this is a quote from an official Zoom video about the magnifier:
“…but it will also raise the volume of the recorded file that you’ll import into the computer.”
https://youtu.be/MsDIGWj4tYo
I have one on preorder from Zoom, looks like B&H is saying March 22…
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I'm assuming that with amplified PA music low levels won't be an issue, but in the case of recording softer material is it better to try to get the volume from the machine while recording (via magnification), or boost in post in terms of noise/artifacts? I'm guessing since it's digital amplification there wouldn't be much difference?
I'm continually curious about this - and have never understood a clear answer. Specifically, if you have good / great preamps is there an expressed benefited to running "hot" (i.e. in 32-bit even if you're clipping here and there just let it go and bring it down in post) or better / just as good to run conservatively and apply a volume boost in a modern DAW in post? I have seen several posts over the years of people positing that their recordings sound better when they are run "hot", sometimes potentially even briefly bouncing off the limiter vs. raising in post - is there any merit to this observation thought / claim? For my part I cannot clearly identify a preference or even a clear difference in cursory A/B testing.
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Let’s really beat this thing to death, this is a quote from an official Zoom video about the magnifier:
“…but it will also raise the volume of the recorded file that you’ll import into the computer.”
https://youtu.be/MsDIGWj4tYo
I have one on preorder from Zoom, looks like B&H is saying March 22…
Well, that statement is still consistent with my earlier speculation that the last waveform magnification set is stored in the extended metadata file - it would only need to store one number in respect of each channel - and your DAW would use that as if the DAW itself had raised the digital gain. They are talking about volume (playback) not level (recording). We'll see...
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In other words, it's like "ReplayGain" in mp3 files. "ReplayGain software stores information in the audio file's metadata header rather than directly affecting the original audio information. This metadata allows audio players and sound systems that support ReplayGain to adjust the volume to the desired level automatically."
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"• Changing the amplification will change the volume and be applied to the recording data." means that the metadata for the recording will include the current amplification so it will play back (in the device at least) at that volume if you close the file and reopen it. I don't know whether DAWs will also pick up that volume data.
I presume this will emulate the behavior of the F8, in which case the recorder remembers internal mixer levels for subsequent playback on the device, but a DAW reading the exported files won't know what the internal mixer levels on the recorder were (the case in my experience)
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I presume this will emulate the behavior of the F8, in which case the recorder remembers internal mixer levels for subsequent playback on the device, but a DAW reading the exported files won't know what the internal mixer levels on the recorder were (the case in my experience)
And if this is how it ends up operating, hitting record is literally the only thing to do? So aside from wanting to see larger waveforms in real time, there would be no reason to use the magnifier/amplifier for the end result?
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That's essentially how I run the F8 now, with input gain settings on the F8 preset.
Even if not visually monitoring peak levels I'd want to see the meters movement to confirm everything is setup and working correctly.
You'll want to set reasonable mixer levels if you monitor with headphones while recording and/or for trouble shooting. I don't do that while recording, relying on visual confirmation by the meters. But it's also nice to have something reasonably close to a good mix dialed in during recording for playing back from the recorder later (because the recorder remembers), even if doing so simply to audibly identify filesets.
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https://www.youtube.com/watch?v=jQgO_FZuya8
Lots of good info here - one key takeaway for our purposes: Looks like about 3hrs @ 32 / 192 w/ P48 on 2 Eneloop Pros. Presumably that will get a bit better with a reduced (i.e normal) sampling rate.
Seems plenty good for our purposes, and you can certainly swap in a couple of spares during a set / artist break.
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By the way, his suggestion here is that the magnification setting does affect levels - he says if you don't set them at the same level when recording, it is a pain to correct in post (rough translation).
https://www.youtube.com/watch?v=k5afJBTb8eI&t=50s
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I know there is a lack of understanding of the terms. On this deck, Gain is set, it cannot be adjusted. That is the incoming signal being recorded. The 2 ADC's in the deck are utilized to extend range so the recording (for the most part and baring a bunch of variables that likely will never happen) will not be distorted.
The output (the recording) is not set. You can raise it (volume) until it distorts, you may have the output set at the start of a recording so loud it distorts or be way too low or anything in between which you can see on the waveform on the screen. and yes you may have set levels so that one channel is louder than the other as you might do on any recording. The good news is whatever program you use when you unload your file will allow you to adjust and change that output (volume a/k/a fader level) exactly where you want it. If it is too loud or distorting, you can lower the levels in post and it will sound fine. If it is too soft, you can raise the levels in post and it will sound just fine. And if one channel is louder or softer than the other you can adjust the levels until they match. This is really no different than any recording or recorder once the recording is complete with the exception your adjustment if too loud, distorted or too low, will sound fine once adjusted which may not be the case on a 24bit or 16bit recording.
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I know there is a lack of understanding of the terms. On this deck, Gain is set, it cannot be adjusted. That is the incoming signal being recorded. The 2 ADC's in the deck are utilized to extend range so the recording (for the most part and baring a bunch of variables that likely will never happen) will not be distorted.
The output (the recording) is not set. You can raise it (volume) until it distorts, you may have the output set at the start of a recording so loud it distorts or be way too low or anything in between which you can see on the waveform on the screen. and yes you may have set levels so that one channel is louder than the other as you might do on any recording. The good news is whatever program you use when you unload your file will allow you to adjust and change that output (volume a/k/a fader level) exactly where you want it. If it is too loud or distorting, you can lower the levels in post and it will sound fine. If it is too soft, you can raise the levels in post and it will sound just fine. And if one channel is louder or softer than the other you can adjust the levels until they match. This is really no different than any recording or recorder once the recording is complete with the exception your adjustment if too loud, distorted or too low, will sound fine once adjusted which may not be the case on a 24bit or 16bit recording.
But if you don't set the levels the same during recording and you then start playing with channel levels independently in post, are you not then potentially damaging the "level difference" aspect of the stereo recording? It seems to me you should be careful with this device to always have the same magnification level set for both channels or you will run into trouble in post.
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The problem is that it has been implied (either in the manual/guide or by user(s), can't remember) that the magnifcation level / replay volume is somehow attached to the file. Or not. And if so, how? And if so, if it is varied during recording, is the variation recorded, or just the final setting (which would be by far the easiest implementatoin of course, just one value for each channel). And if it's a metadata thing, will DAWs read it at all anyway? And if the magnification is varied during replay in the device after recording, is that new value saved in the same way? 'Cos if so, before editing, you could reset it to zero for each channel in the recorder, end of any problem when editing. So - there are still quite a range of possiblities (and problems, potentially) with this magnification feature.
As for stereo image, no, varying channel levels is essentially no different to using a balance control.
Generally, it has to be remembered that live music recording may not be the prime target of this device - it may have two channels as much for two lav mics being worn by two people, on soft spoken and the other not, as for a natural stereo recording.
[Edit - ah I've seen the catch - there's no adjustment for magnification during playback on the device. That's why you need to be careful with it. But if your DAW doesn't respect the stored magnifcation volume, it doesn't really matter anyway.]
[Another edit - actually, it's possible that the magnification level doesn't get stored as such at all - it could be simply digital playback/monitoring volume control - the wording of the manuals could be read that way, if you take into account translation issues.]
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Just got an email in stock alert from Sweetwater but I see no way to add it to cart :shrug: Currently stating 3 available...
https://www.sweetwater.com/store/detail/F3Zoom--zoom-f3-2-input-field-recorder
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Just got an email in stock alert from Sweetwater but I see no way to add it to cart :shrug: Currently stating 3 available...
https://www.sweetwater.com/store/detail/F3Zoom--zoom-f3-2-input-field-recorder
I have always had the best experiences with Sweetwater when I spoke directly to one of their sales reps via phone or email.
Jason Koons (https://www.sweetwater.com/jason_koons) (who I have bought many things from over the years) was able to get me an F6 when they were newly arriving but showing out of stock everywhere, including on SW's website.
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Just got an email in stock alert from Sweetwater but I see no way to add it to cart :shrug: Currently stating 3 available...
https://www.sweetwater.com/store/detail/F3Zoom--zoom-f3-2-input-field-recorder
I have always had the best experiences with Sweetwater when I spoke directly to one of their sales reps via phone or email.
Jason Koons (https://www.sweetwater.com/jason_koons) (who I have bought many things from over the years) was able to get me an F6 when they were newly arriving but showing out of stock everywhere, including on SW's website.
Thanks, good to know. My Zoom order shipped and is a week away! I’ll update with any findings; not even sure if my current editing program (Amadeus Pro) does 32-bit float :hmmm:
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Excited to see how this thing works! Great potential as a set it and forget it device (on stage, at the board, etc.)
NOT trying to get into the 32bit debate with that comment…
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Excited to see how this thing works! Great potential as a set it and forget it device (on stage, at the board, etc.)
NOT trying to get into the 32bit debate with that comment…
The only debate would be with people who haven't used multi-ADC 32FP recording. I have had an F6 since the early days of its release and only ever use 32FP mode. The "set and forget aspect" has totally proven itself. I record music with much wider dynamic range than most members here, and there have been zero audible artifacts of the auto-ranging ADCs switching. It is truly transparent.
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Or with people who realize it is not necessary if you figure out the relationship between your mics sensitivity and the maximum input of the recorder. If you like it, great. But it is not hard to put yourself in an "impossible to overload unless you overload the mics or recorder input" situation. It is interesting to note that the pro recorders, like the SD Scorpio/888/833 or Zaxcom Nomad, don't offer it. Those recorders also offer a pretty huge dynamic range, with dual (or more) ADCs, and they are basically impossible to overload if you take a minute to figure it out. Very simple with a recorder with a high input level...
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Recording acoustic music, I gradually became aware that regardless of the size of the musical group involved, eg symphony orchestra down to solo acoustic guitar, I was always setting the gain control on my M/S mic preamp to the same level every time, and likewise the recorder. Being a bear of very little brain, I marked these points with a big red line on the knobs'scales. To compensate for the very different level emitted by an orchestra vs a guitar, it was a matter of being much closer to the guitar than to the orchestra - which was the natural thing to do with the M/S mic pair anyway. So my theory was that there is only one level of sound (with its attendant dynamic range) in acoustic music, and so there's only one level to set with the equipment. No 32 bit float required.
Except... sometimes there were exceptions. And you couldn't always be sure when an exception would strike and you'd see full scale on the meters, oops. One example might be when an orchestra was fortissimo in a reverberant space, and you'd get a build up of reverb plus direct which would sum together to make the meters melt. 32 bit float would have been a source of comfort on those occasions. But having said that, off the top of my head I don't recall a single instance where a live recording was seriously affected by incorrect level setting. But it's a stress relief thing to have 32 bit float.
The other stress relief thing was pointed out to me by a well known classical musician. "Peter, you should always arrive an hour earlier than you need to at every concert. That way you will have lees stress and you will live longer." Unfortunately he was well known for turning up for concerts at the last minute.... but I think he's still alive...
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Except... sometimes there were exceptions. And you couldn't always be sure when an exception would strike and you'd see full scale on the meters, oops. One example might be when an orchestra was fortissimo in a reverberant space, and you'd get a build up of reverb plus direct which would sum together to make the meters melt. 32 bit float would have been a source of comfort on those occasions. But having said that, off the top of my head I don't recall a single instance where a live recording was seriously affected by incorrect level setting. But it's a stress relief thing to have 32 bit float.
The problem, as I understand it, is that 32-bit float probably wouldn't save you in that situation. The "magic" non-clippable aspect applies to the digital domain; if you clip in the analog stage, it's just clipped. But the dual ADCs, and the huge dynamic range they impart, let you be a little more conservative in your analog gain setting, which hopefully helps with the exceptions.
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Experiencing clipping at the mic itself or at a sensibly configured preamp wouldn't be likely in recording acoustic music with appropriately placed mics (even in a reverberant acoustic). That's my point - in that kind of recording, the initial moderator of the level is the source-to-mic distance. But indeed, other sound sources are a different kettle of fish.
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^ Yes, I agree. The distance is really crucial. The only time I have managed to clip anything, really, was on-stage a little too close to Dave King, who can be a force of nature...
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Everything evolves. I've been taping since 1971. I remember the move to a tiny Walkman D6. Wow, it was the greatest. I just have to chuckle when I read arguments against not having to worry about clipping. It's a huge advantage whether you need it or not. Personally I think it's fantastic, and I never worried that I'd screw up my levels. It's still fantastic. I love my MixPre 6II and Zoom F6 decks. Plus it's coming like it or not (what's not to like? ) . I have not heard a single argument to date against it that sounds rational.
Reminds me of the chatter when Flash replaced DAT. ;D
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^ Yes, I agree. The distance is really crucial. The only time I have managed to clip anything, really, was on-stage a little too close to Dave King, who can be a force of nature...\
As you mention, I've found on-stage drums to be what reaches and defines the practical SPL limit of my current F8-based rig. On stage last Saturday night, or rather on-floor (no actual stage) up close to the band with my mics in an on-stage position necessary due to the highly reverberant nature of the performance space, drums clipped the input stage of one channel into the F8. Input trim was turned all the way down, advanced limiting turned on. Only happened once or twice over the course of the evening I think (had metering set to hold any input overs) and it isn't audible. This mic-sensitivity input level limit can be avoided with the F8n, in which case I could switch input sensitivity from mic to line-input sensitivity and achieve greater headroom while retaining phantom power.
Knowing where that limit lies is the key. Recording 24 bits, I mostly run F8 in this kind of set and forget 32-bit float type of way. Sometimes I'll pull up the input gains across all channels for quiet stuff, not for but mostly just to get more level so I can see all the meters moving, including the rear-facing channels
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^^ I guess you're referring to me, but that's not really what I am saying. I have nothing against 32BF, but I realized that it is more complicated than the marketing stuff portrays. The "no need to set levels" is due to the multi-ADC architecture and the huge dynamic range that results. That's been around for a while, I think, but not in lower-priced stuff. I don't think I have changed the gain setting on my MixPre-6 (first version) in ages. No need, really. If I peak at -6 or -12 or even -22, it sounds pretty much the same once boosted in the DAW. The non-clippability is different. The Dave King recording I mentioned would virtually certainly have overloaded the F3 due to the sensitive mics (DPA 2006C; 40 mV/Pa) and being too close to a fired-up King!
As you mention, I've found on-stage drums to be what reaches and defines the practical SPL limit of my current F8-based rig.
My last last few on-stage recordings, I cheated towards the other side of the stage a bit. More balanced that way, actually, when the drummer is hitting hard.
Knowing where that limit lies is the key. Recording 24 bits, I mostly run F8 in this kind of set and forget 32-bit float type of way. Sometimes I'll pull up the input gains across all channels for quiet stuff, not for but mostly just to get more level so I can see all the meters moving, including the rear-facing channels
That's basically what I was trying to say above...
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My last last few on-stage recordings, I cheated towards the other side of the stage a bit. More balanced that way, actually, when the drummer is hitting hard.
Love running mics on stage. Not too close, not too far. There is no greater exciting realism for drums. Regardless of mic'ing distance, I find a straight line-of-sight from mics to the snare to be a good rule of thumb,
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I received my unit today... all view magnification changes DO NOT effect the waveform
I ran some tests with various inputs and will have some shows coming up in a few
I plan on running one of my various preamps in front since it is so TINY
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Beat me! Awesome news keep us updated on your experiences :cheers:
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I received my unit today... all view magnification changes DO NOT effect the waveform
I ran some tests with various inputs and will have some shows coming up in a few
I plan on running one of my various preamps in front since it is so TINY
So, in essence, magnification is simply digital monitoring level? And does the magnification setting carry over to replay - in the machine, not n the DAW - or does it only function while recording? Looking forward to your updates! I'd love to get one simply because it would represent to me the final stage in a very long audio recording journey (my first pro recording was in 1968!) , but I guess I'll leave it to the younger folks to carry the flag now!
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Excited to see how this thing works! Great potential as a set it and forget it device (on stage, at the board, etc.)
NOT trying to get into the 32bit debate with that comment…
The only debate would be with people who haven't used multi-ADC 32FP recording. I have had an F6 since the early days of its release and only ever use 32FP mode. The "set and forget aspect" has totally proven itself. I record music with much wider dynamic range than most members here, and there have been zero audible artifacts of the auto-ranging ADCs switching. It is truly transparent.
I should clarify my earlier post:
I was not taking issue with people who choose not to use this recording method, nor was I saying it solves all of your problems. You can still overload the input stage, as Ozpeter and aaronj point out.
The point I was trying to make is that the technology works and does so seamlessly, and that the concerns expressed on this board about detrimental effects / artifacts have not borne out in practice.
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I received my unit today... all view magnification changes DO NOT effect the waveform
I ran some tests with various inputs and will have some shows coming up in a few
I plan on running one of my various preamps in front since it is so TINY
Thanks for reporting back. Wow, that is a small device!
Side question: Did you make those side-exit XLRs yourself, or were they a custom build by a vendor?
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I received my unit today... all view magnification changes DO NOT effect the waveform
I ran some tests with various inputs and will have some shows coming up in a few
I plan on running one of my various preamps in front since it is so TINY
Thanks for reporting back. Wow, that is a small device!
Side question: Did you make those side-exit XLRs yourself, or were they a custom build by a vendor?
I spent TOO much and purchased them... they are adjustable
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Side question: Did you make those side-exit XLRs yourself, or were they a custom build by a vendor?
I spent TOO much and purchased them... they are adjustable
Very sleek! Are these the brand?
https://www.bhphotovideo.com/c/product/1296365-REG/cable_techniques_ct_lpxr_10r_low_profile_xlr_3_pin_10.html
Looks like they distribute parts/kits through Redding Audio for the more electronically industrious amongst us:
https://www.cabletechniques.com
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In this video - https://youtu.be/zpoFhCCi2VE - it's stated that magnification level changes during recording do not affect the recorded file, but that magnification changes set before recording DO affect the recorded file - if that's the case doesn't that negate the object of using 32 bit float (ie no need to have recording level setting)? Or maybe the idea is that 32 bit float has such a vast headroom that even upping the recorded level to the maximum you still won't get digital clipping (if you bring the level down in your DAW).... help! I'm confused...
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Or maybe the idea is that 32 bit float has such a vast headroom that even upping the recorded level to the maximum you still won't get digital clipping (if you bring the level down in your DAW).... help! I'm confused...
This is my understanding, and hope; and that it works the other way as well for noise floor on a file recorded “too” low.
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Or maybe the idea is that 32 bit float has such a vast headroom that even upping the recorded level to the maximum you still won't get digital clipping (if you bring the level down in your DAW).... help! I'm confused...
This is my understanding, and hope; and that it works the other way as well for noise floor on a file recorded “too” low.
On my F6, if I set the channel knobs to REC level, I can crank them all the way and record files that have ridiculously high levels - we're talking +50 dB or more. Once I copy those files over and import into a wave editor, those levels can be brought down under 0 dBFS and there is no clipping.
My general habit now is to record with the channel levels set at +20 dB, only because I find I have to adjust the overall level less in post. It really doesn't matter though.
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Received my unit yesterday!
First off, to confirm once and for all:
Magnification settings done before starting a recording DO affect the level of the .wav brought into your DAW
It also saves your magnification settings on each channel after power down and swapping batteries, that's nice. It is incredibly small in person, more so even than in pictures. It's a solid mass and feels well built. It sits perched sturdily on a desk for those using it as an interface. The battery door is a bit thin and doesn't instill confidence in longevity but so are the ones on my Tascams and I haven't had a problem with those in 10 years. One thing I do not like is that after starting a recording, if you slide the record button up to record again, it stops the recording. Not a huge deal but the switch does move very easily and I can't imagine why it functions this way. I believe on most recorders pressing record again makes a new track or marks the file. What reason would you ever have to stop a recording so quickly that you can't take the time to hit the actual stop button--besides wanting either of the aforementioned actions? Even though you should be using the hold function, I still see no reason for that, it's just a liability.
I did a test recording some music and then room silence on the minimum and maximum magnification settings. The quieter file peaked at -43dB and the louder file +16dB. I normalized each to -.1dB and then listened to the silent portions. They were nearly indistinguishable, and I'm sure that in a controlled environment the noise floors would sound exactly the same.
For my Energizer E2 battery runtime test, I set it to 48kHz, lowered the headphone volume to zero (nothing connected), left the magnification to the stock 16x on both channels and ran 2 CCM 4 mics with 48V phantom power. Total record time was 6:37, which I'm very pleased with. That's about an hour less than my current recorders but with one less battery.
This is the perfect box for no fuss 2 channel XLR recording. I will likely pick up another for when I do matrix shows. Look forward to running it out in the field soon. Any other questions that can be answered just ask 8)
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Received my unit yesterday!
First off, to confirm once and for all:
Magnification settings done before starting a recording DO affect the level of the .wav brought into your DAW
It also saves your magnification settings on each channel after power down and swapping batteries, that's nice. It is incredibly small in person, more so even than in pictures. It's a solid mass and feels well built. It sits perched sturdily on a desk for those using it as an interface. The battery door is a bit thin and doesn't instill confidence in longevity but so are the ones on my Tascams and I haven't had a problem with those in 10 years. One thing I do not
like is that after starting a recording, if you slide the record button up to record again, it stops the recording ??? Not a huge deal but the switch does move very easily and I can't imagine why it functions this way. I believe on most recorders pressing record again makes a new track or marks the file. What reason would you ever have to stop a recording so quickly that you can't take the time to hit the actual stop button--besides wanting either of the aforementioned actions? Even though you should be using the hold function, I still see no reason for that, it's just a liability.
I did a test recording some music and then room silence on the minimum and maximum magnification settings. The quieter file peaked at -43dB and the louder file +16dB. I normalized each to -.1dB and then listened to the silent portions. They were nearly indistinguishable, and I'm sure that in a controlled environment the noise floors would sound exactly the same.
For my Energizer E2 battery runtime test, I set it to 48kHz, lowered the headphone volume to zero (nothing connected), left the magnification to the stock 16x on both channels and ran 2 CCM 4 mics with 48V phantom power. Total record time was 6:37, which I'm very pleased with. That's about an hour less than my current recorders but with one less battery.
This is the perfect box for no fuss 2 channel XLR recording. I will likely pick up another for when I do matrix shows. Look forward to running it out in the field soon. Any other questions that can be answered just ask 8)
+t thank you for this review! Answered many questions...great stuff!
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> One thing I do not like is that after starting a recording, if you slide the record button up to record again, it stops the recording. Not a huge deal but the switch does move very easily and I can't imagine why it functions this way.
This sounds rather bleak, actually.
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I guess that it will be necessary to always slide one way to record, and then the other way immediately to hold. Maybe they thought that an accidental press of the stop button would be more likely compared to an accidental slide of the record control. But it does seem a bit risky...
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Check the system menu structure to see if there is a setting for what happens when pressing (or sliding) record while already recording.
Just thinking it might be similar to F8 where the Play button can be set to pause or mark, or pause + mark, as well as being set separately for recording and playback. I think the Record button function can be modified as well, maybe starting a new recording verses pausing or something, but I can't recall the details of that one.
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Check the system menu structure to see if there is a setting for what happens when pressing (or sliding) record while already recording.
Just thinking it might be similar to F8 where the Play button can be set to pause or mark, or pause + mark, as well as being set separately for recording and playback. I think the Record button function can be modified as well, maybe starting a new recording verses pausing or something, but I can't recall the details of that one.
I don’t see that there is unfortunately. Is this something that firmware is able to address?
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finally got around to using it last night
I set the the display for 4x on each channel for the opener and I ended up going OVER by about 2dB, but it was fine after lowering in post
for the main act, I ran the the display at 1x and had to raise levels in post about 5dB (Schoeps mk22/NBobs > NBox Elite)
SO THE DISPLAY IS BASICALLY A VOLUME CONTROL
I really like the ease of operation and the small size, plus it didn't use too many lights on my ANKER
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Good to hear; yes the simplicity and size are quite appealing, thanks for the update!
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How’s the A/D converter on the F3?
finally got around to using it last night
I set the the display for 4x on each channel for the opener and I ended up going OVER by about 2dB, but it was fine after lowering in post
for the main act, I ran the the display at 1x and had to raise levels in post about 5dB (Schoeps mk22/NBobs > NBox Elite)
SO THE DISPLAY IS BASICALLY A VOLUME CONTROL
I really like the ease of operation and the small size, plus it didn't use too many lights on my ANKER
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finally got around to using it last night
I set the the display for 4x on each channel for the opener and I ended up going OVER by about 2dB, but it was fine after lowering in post
for the main act, I ran the the display at 1x and had to raise levels in post about 5dB (Schoeps mk22/NBobs > NBox Elite)
SO THE DISPLAY IS BASICALLY A VOLUME CONTROL
I really like the ease of operation and the small size, plus it didn't use too many lights on my ANKER
So my interpretation of the block diagram was correct; the F3 "waveform amplification" works just like the F6 when track knobs are set to REC LEVEL mode.
Thanks for the report! I'm thinking if all goes well, the F3 may become a popular recommendation for a compact one-box recording setup.
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Mine turned up from B&H yesterday. As mentioned previously, it is significantly smaller than the pictures made me think it would be. Setup was simple / did not require manual. Build quality is about on par with similar devices in this price range, although as mentioned, the battery door does feel rickety / cheap like you'd find on a child's toy.
I'm going to try to get out to a small jazz club this weekend with either Line Audio CM4 or OM1 fed directly into the F3 and will share a sample when available.
I am particularly curious about how "good" it will sound as compared to what I get out of CM4 direct XLR in to my MixPre3-ii - which I love. For anyone with experience with the Zooms, any predictions? I anticipate it is going to be a bit more bright / harsh and not as warm, which seems to be a unique quality of the MixPre line as compared to my previous experiences with Zoom devices. Obviously there will obviously be a bit more preamp noise, but I'm guessing that will be largely insignificant in our applications.
I do note that some are using these line-in with a USBPre providing amplification. To me this significantly reduces the utility of the device - if I need to have something like a USBPre in front I'd just as soon use my MixPre3-ii instead as it is a smaller footprint and not any better from a security perspective.
Will report back soon.
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Mine turned up from B&H yesterday. As mentioned previously, it is significantly smaller than the pictures made me think it would be. Setup was simple / did not require manual. Build quality is about on par with similar devices in this price range, although as mentioned, the battery door does feel rickety / cheap like you'd find on a child's toy.
I'm going to try to get out to a small jazz club this weekend with either Line Audio CM4 or OM1 fed directly into the F3 and will share a sample when available.
I am particularly curious about how "good" it will sound as compared to what I get out of CM4 direct XLR in to my MixPre3-ii - which I love. For anyone with experience with the Zooms, any predictions? I anticipate it is going to be a bit more bright / harsh and not as warm, which seems to be a unique quality of the MixPre line as compared to my previous experiences with Zoom devices. Obviously there will obviously be a bit more preamp noise, but I'm guessing that will be largely insignificant in our applications.
I do note that some are using these line-in with a USBPre providing amplification. To me this significantly reduces the utility of the device - if I need to have something like a USBPre in front I'd just as soon use my MixPre3-ii instead as it is a smaller footprint and not any better from a security perspective.
Will report back soon.
Looking forward to your report. If it uses the same preamps as the F6, I would be surprised if you found it to be bright or harsh. I record with my CM3s direct into my F6 more than any other combination. I don't have a MixPre-II to compare against, though. I agree that using an external preamp with this kind of defeats the purpose.
As for preamp noise, I doubt you'd notice any. Gain on the F3 is fixed at a fairly low level (again, assuming the F6 preamp design is followed). Your mic self-noise would almost certainly be higher than any noise from the preamp.
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Had my first time out with the device last weekend. >:D in a very small jazz hall (An Die Musik in Baltimore). Band was the Paul Carr Legacy Quartet.
Recorded with Line Audio OM1 (48V) in Healy Technique, mounted to a stationary >:D "object" @ DFC, approximately 2 feet away from stage lip and about 3 ft high with no obstructions. Show was about an hour and a half and I appeared to have about 50% battery left when done (but recording was inadvertently stopped for a while, see point below).
Overall experience with the device was good, however the slide start / stop record thing is decidedly poor. Especially in a >:D situation, given the very minor amount of force required to activate the slide function, there is a significant potential for a minor bump or even cables settling in a bag to stop recording inadvertently. This occurred during this outing, and I lost a portion of the show on this device as a result. Not a fan. A simple update requiring a 2 second hold slide would greatly reduce what seems to be a pretty significant risk to me (have a high number of >:D shows under my belt and have a well worked out "system" - in spite of that this happened to me on the first outing). I can't recall the last time I've had an issue with a recorder being stopped when I was able to check it (other than perhaps a dead battery).
Re: Tone - To my ear it is definitely brighter and lacks some of the warmth I am used to in this hall with these mics wired into a SD MixPre3/6-ii. I would say there is plenty of bass there and the recording is pleasing enough, but something is off and weighted toward what I would describe as bright as compared to several previous recordings with the same rig into the MixPre. Certainly "correctable" with EQ in post but it does reinforce my affection for the "out of the box" SD sound. Obviously I need some more shows under my belt with the device to corroborate these initial thoughts. These things are hard to nail down - it could be entirely placebo effect.
As mentioned previously by myself and others, the size of this thing is tiny and this is its biggest asset. Since I purchased my first fully digital recorder (Sony PCM-D1) I have been longing for something very small, with no (inevitably poor) internal mics adding to the footprint, and including XLR connectivity as I have always loved the simplicity of a single reliable cable in between the each mic and the recorder. Lately, after using other 32-bit devices, this requirement has also made it on to my list. Especially for >:D or even semi- >:D the ability to completely ignore gain settings on the device has been a game changer for both me and my setup / ease of deployment. With those requirements in mind, this device comes closest to a realization of that dream, and as a result I will likely be using this for any place where >:D and security is a concern.
At some other venues I frequent where it's not explicitly open taping but more of a "be discrete" policy, I think I would still be more likely to bring out the SD as it is more tonally pleasing direct from the device (to my taste) and the slight increase in size doesn't matter to me as I'm not trying to crotch the thing.
I'll report back again after another go at it this weekend in a large opera hall setting from FOB with the CM4s. Eager to see how it performs with those mics as well.
Hope this is useful others. Let me know if you have any questions. Raw from device WAV sample of a track can be found grabbed here for a week: https://we.tl/t-1IxD6OpCMm
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I look forward to hearing this sample later, thanks for posting that and your account of your real-world experience. I suspect that Zoom did not think in terms of this kind of taping when designing the device and that's why the obvious problems with the record slider (in that situation) were not encountered (presumably) in their testing. Did you have the switch in the "hold" position, meaning that it was pushed from "hold" all the way to "stop recording?" Without consulting photos of the device I strongly suspect that someone will come out with a third party physical device to protect the switch from inadvertent operation - or that they will fix the problem with a firmware update (for instance, requiring a two button operation to stop - a long hold of the button might well not have fixed the problem in the bag you encountered if indeed a cable was resting on the switch).
[Edited to add that I'm listening to the sample right now - nice! - but I felt the need to lift the upper frequencies somewhat, as the cymbal sound in particular sounded a bit veiled to me. But I'm listening with 73 year old ears... yesterday they were only 72 years old... sigh!]
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Thanks for the feedback / thoughts / ideas. It's my pleasure to share my experience, glad it's helpful.
Few points / thoughts:
- The record switch was locked when I placed the device. It doesn't take much force at all to disengage the lock.
- Good point about the long hold being shortsighted - it is possible that the cable scenario would still cause the recording to stop.
- I'm not sure what happened but I would suspect that after deploying said object something settled just the right way to slide the switch.
- I did buy the BT module for connecting to phone / monitoring - this would have helped catch this quickly but unfortunately the previously promised March launch for the Android software appears to be vaporware at this point - now saying "Spring 2022"
- Interesting feedback on the equalization - can't say it was unexpected. There are so many variables affecting what is heard (not the least of which is the playback hardware), so it's certainly within reason that your observation is the exact opposite of mine! One more reminder not to get too heavy handed with EQ and stay more purist, I suppose.
Lastly, a belated Happy Birthday to you! I'm hoping I still have ears to speak of by the time I am 73! Glad to see you're still in the game.
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"given the very minor amount of force required to activate the slide function, there is a significant potential for a minor bump or even cables settling in a bag to stop recording inadvertently."
We are not target audience but this design flaw is universal and inexcusable.
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"given the very minor amount of force required to activate the slide function, there is a significant potential for a minor bump or even cables settling in a bag to stop recording inadvertently."
We are not target audience but this design flaw is universal and inexcusable.
It's particularly frustrating that they felt the need to "improve" an existing, field proven, verified working design. Why the typical approach of a slightly recessed record button and a dedicated lock switch needed a redesign is not clear to me and worse, nothing seems to have been made better in the process. The German notion of "schlimmbesserung" comes to straight to mind.
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Thanks for the feedback / thoughts / ideas. It's my pleasure to share my experience, glad it's helpful.
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Lastly, a belated Happy Birthday to you! I'm hoping I still have ears to speak of by the time I am 73! Glad to see you're still in the game.
Thanks! I'm chiefly into video these days. But the F3 tempted me until I saw that it doesn't support M/S monitoring. I do have the F1 - that has an option to lock the buttons during recording, and you have to press "stop" and "record" at the same time to stop recording and unlock. I would have thought that would be the way to fix this problem on the F3 (presumably it doesn't have that option hidden in the menus somewhere? Given its family resemblance to the F1, it would have been the obvious thing to do).
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I liked the F3 because it would "move" view during playback.That is, if you record ambisonic, you can move the unit while it is playing, and it will move the listener.Pretty cool interface.
Unfortunately, the USB connector on it broke. Very unhappy with Zoom. Great projects, but poor quality control and design.
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[Edited to add that I'm listening to the sample right now - nice! - but I felt the need to lift the upper frequencies somewhat, as the cymbal sound in particular sounded a bit veiled to me. But I'm listening with 73 year old ears... yesterday they were only 72 years old... sigh!]
Happy belated birthday from San Francisco! :-)
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Happy belated birthday from San Francisco! :-)
Thanks.... SF... I haven't been back there since 1968... those were the days. Somewhere I have a reel-to-reel recording of an interview I did with a hippie drug addict there (charming young lady actually) but heaven knows where I've put it!
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I think the best way to fix this hold/record switch issue is through a firmware update that does the following:
1. REC/HOLD switch only starts recording. Engaging it again after rolling only add markers (or does nothing at all).
2. To stop recording, the HOLD must be disengaged, and then you must do a 2-second press on the STOP button.
These changes would compensate for the relatively low disengaging force of the hold switch, move the stop function to another switch entirely, and prevent accidental stops.
Once, again, they had all of this figured out quite well on the F6: REC just records. STOP stops. The hold function is engaged by a 2-second press on the headphone volume control. They could have done all of that exactly the same on the F3 and saved some money by having a simple button instead of a two-position slider.
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Second outing report:
Gaffer's taped the switch in the lock position this time - not going to be fooled again!
This time it was Joey DeFrancesco @ Berk's Jazz Fest using Line Audio CM4 (48v) from FOB, DFC ~Wide ORTF mounted to a stationary >:D "object" direct into the F3 @ 32, 48. 1.5 hour show still left me with about half battery using 2 Eneloop Pros - the battery life on this is also turning out to be more than sufficient which is very nice.
Size of device was once again its best asset - was not at all of interest during bag search. :)
Seems more weighted toward "warm" this time, but I still have the impression that similar outings with the SDs just sound better out of the device. With that said I'm sure I could EQ to taste for a nearly identical result. Don't hear anything fundamentally wrong with the file. Perhaps I've just had too much of the SD Kool-aid at this point. I have a couple pairs of these mics so the next chance I get I'd like to compare the 2 recorders with identical mic setup to see if there's something there or it's all in my head.
Sample track available here for a week: https://we.tl/t-dfMx2s8DUP
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Sample track available here for a week
I like it, but when the synth player plays, it drowns out Joey's B3.
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3rd outing:
The F3 is continuing to be a winner. The gaff tape, while not ideal, is proving to be effective in avoiding erroneous stops. Battery life continues to be great with the Eneloop Pros - I am never below 50% by the end of a ~ 1.5 hour set @ 32/48 w/ 48v phantom on. Bag checks don't even see the Hermitshell case I have the F3 in opened up - it is small enough that it is entirely disregarded by security so far - obviously that will vary by venue. I'm going to do a Herbie show soonish - he tends to have much tighter screening from experience, so it will likely get its real test then.
In terms of ease of stealth operation, it really doesn't get any simpler. I am eyeballing levels based on house music before hitting record and dumping it till the end of show - not checking anything but an occasional glance at the red recording light (out of paranoia from the past lost recording on the first outing). Results have been great so far and the 32-bit would make even significantly botched levels a non-issue.
This time there was far too much bass so I've done some minor post to the samples as it was not listenable as it was. Still too much cymbal/hh "crashiness" going on for my taste so I did minor subtractive EQ on that as well.
All things considered, after 3 outings with different types of venues / different mic setups / etc I am giving this one a big thumbs up. As mentioned previously I am inclined to use the MixPre3-ii whenever there is no concern for security rummaging through the bag, based on my subjective assessment that it just sounds "better" raw from device, but when stealth is a factor I no longer feel like I am sacrificing as much as I did with the stealth oriented Sony A10, D50, D1, M10 / Sony XLR-1 / Battery Box setups I was using previously.
Recorded directly behind soundboard using a slightly narrow (practical stealth issues) NOS on stealth "object" peeking over the rail ~ 6ft high (very low "balcony") @ DFC, FOB with Line Audio CM4 > F3.
Artist: Keiko Matsui
Venue: Bethesda Blues and Jazz Club, Bethesda, MD.
Sample 1:
https://we.tl/t-Nc0HP9SPXl
Sample 2:
https://we.tl/t-kxPahRPcVZ
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3rd outing:
The F3 is continuing to be a winner. The gaff tape, while not ideal, is proving to be effective in avoiding erroneous stops. Battery life continues to be great with the Eneloop Pros - I am never below 50% by the end of a ~ 1.5 hour set @ 32/48 w/ 48v phantom on. Bag checks don't even see the Hermitshell case I have the F3 in opened up - it is small enough that it is entirely disregarded by security so far - obviously that will vary by venue. I'm going to do a Herbie show soonish - he tends to have much tighter screening from experience, so it will likely get its real test then.
In terms of ease of stealth operation, it really doesn't get any simpler. I am eyeballing levels based on house music before hitting record and dumping it till the end of show - not checking anything but an occasional glance at the red recording light (out of paranoia from the past lost recording on the first outing). Results have been great so far and the 32-bit would make even significantly botched levels a non-issue.
This time there was far too much bass so I've done some minor post to the samples as it was not listenable as it was. Still too much cymbal/hh "crashiness" going on for my taste so I did minor subtractive EQ on that as well.
All things considered, after 3 outings with different types of venues / different mic setups / etc I am giving this one a big thumbs up. As mentioned previously I am inclined to use the MixPre3-ii whenever there is no concern for security rummaging through the bag, based on my subjective assessment that it just sounds "better" raw from device, but when stealth is a factor I no longer feel like I am sacrificing as much as I did with the stealth oriented Sony A10, D50, D1, M10 / Sony XLR-1 / Battery Box setups I was using previously.
Recorded directly behind soundboard using a slightly narrow (practical stealth issues) NOS on stealth "object" peeking over the rail ~ 6ft high (very low "balcony") @ DFC, FOB with Line Audio CM4 > F3.
Artist: Keiko Matsui
Venue: Bethesda Blues and Jazz Club, Bethesda, MD.
Sample 1:
https://we.tl/t-Nc0HP9SPXl
Sample 2:
https://we.tl/t-kxPahRPcVZ
Excellent write up! Definitely see the potential for one of these…
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I love my F6. I want one of these too! It's like secret weapon.
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Ordered one…
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End user report is eagerly awaited! I hope you get delivery soon.
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Couple of questions
-Anyone use the bluetooth adapter/app? Based on the issues with the recording slide, seems like gaff-taping that in place and using the app to control makes some sense?
-Thoughts/ideas on mounting it? It primarily won't be a >:D tool for me, and I'm thinking I'd like to just mount it right to the stand. Could probably use those bars and tape it, but some sort of clamp option would be nice too.
-Has anyone tested it using AA batteries, plus an external pack? Curious if it's a seamless transition if the external pack dies (my understanding is that it defaults to external power first). I'd like to use a smallish usb battery with it if possible.
Should be getting it Friday, probably running it for a festival in a few weeks!
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Couple of questions
-Thoughts/ideas on mounting it? It primarily won't be a >:D tool for me, and I'm thinking I'd like to just mount it right to the stand. Could probably use those bars and tape it, but some sort of clamp option would be nice too.
https://www.bhphotovideo.com/c/product/1422139-REG/smallrig_2164_multi_functional_crab_shaped_clamp.html
I use this for my A10, should work nicely with the F3.
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I like that clamp a lot, and the price is great too.
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Couple of questions
-Thoughts/ideas on mounting it? It primarily won't be a >:D tool for me, and I'm thinking I'd like to just mount it right to the stand. Could probably use those bars and tape it, but some sort of clamp option would be nice too.
https://www.bhphotovideo.com/c/product/1422139-REG/smallrig_2164_multi_functional_crab_shaped_clamp.html
I use this for my A10, should work nicely with the F3.
Perfect - exactly what I'm looking for! Thanks!
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Okay, I am now a convert to 32 bit floating. Preliminary impressions, not strict comparisons or measurements.
I had used both a mixpre 6 II and a Zoom F6 for recording chamber concerts before Covid. Since I could always set levels and get enough gain from those pre-amps I didn't really find too much of an advantage from recording 32 bit v. 24 bit on these recorders or v. 24 bits on the F8n. I could always add 8-10 dB in post without much impact on the final product, and set levels conservatively.
But I also use an MMA-A plus iPhone with Metarecorder, a major drawback of this setup in some cases was the limitation of the MMA-A to 30 dB gain. With DPA 4060 mics this has never been a problem, but I also use the DPA supercardioid 4081s (with a bass boost in post) for some things, and these mics are less sensitive. For most louder things, a boost in post of 8-12 dB would be sufficient and did not spoil things; but very quiet programs like harpsichords or viols from a distance in a large hall begged for 20-30 dB additional boost in post, and that was generally ugly. I eased into feeding the iPhone from a Sonosax SX-M2D2, but using only a cautious 35-40 dB of gain the needed boost in post still didn't sound good. Since the Sonosax is a fantastic preamp A/D, I was disappointed that the necessary gain in post was still problematic. Last week I got the settings right, a total of 57 dB (20 "pre-gain" + 37) got very nice results. So today I went with a Zoom F3 alone, DPA4081s and DAD6001s to use the F3 phantom power to run the mics. The resulting 32 bit file was a distinct improvement, seems almost as good as the M2D2 with levels set correctly. YMMV, but I'm impressed with the Zoom implementation of 32 bit and look forward to the format in even slimmer machines than the F3.
Now these comparisons are only impressionistic, different groups played, different repertory. But it's in the same space and I'm in the same seat. I've had some intermittent issues with Metarecorder lately, including a complete loss tonight (I think it's a stability issue with newer iOs upgrades, happens with different MMA-A units and mics and went away with deleting and reloading Metarecorder). So I hope to try the F3 at other places, also with 4060 mics.
Jeff
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Jeff - Thanks for your detailed report. I was particularly impressed to hear you say that your results with the F3 were close to those using the M2D2, which is considered an elite-class pre/ADC. You should share that with some of the people on the GS Remote forum who think Zoom products can never produce pro-level recordings.
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results with the F3 were close to those using the M2D2, which is considered an elite-class pre/ADC.
Only with respect to what is a somewhat distant recording (15-20 feet) in a non-ideal space. In a more ideal setting, the M2D2 blows away most competition. What I think I found is that where I need 50-60 dB of gain the F3 is better than anything that will only give me 30 dB. But even saying this on GS will have Plush dumping on me, whether he's heard an F3 or not.
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So Plush is still at it? As I recall it he was significantly the reason why I gave up posting there. Commonsense views just didn't seem to be respected there. Well, that's insulting the other members, but you know what I mean, probably...
Some years ago I worked in classical music live recording and broadcasting. The radio station moved to new premises, and a small studio was built, big enough for a baby grand and a string quartet, for instance. In the main control room, there were some exotic preamps installed, I honestly forget the maker. Anyway, due to circumstances which I also now cannot recall, we had to route the audio from those preamps via a humble 4 channel Behringer mixer. It was not uncommon to hear people gush on about the amazing sound from the upmarket preamps. It there was a strict rule that nobody was told about the downstream Behringer mixer, which was hidden in some cabinetry.
Now I'm not saying that there's no point in using top flight preamps etc etc. I'm just saying that meaningful comparisons can be difficult to make once you get to a certain level. Still, as a rule of thumb when selecting (say) a new pair of speakers in a hifi shop, first ask to hear your test material on whatever speakers they have on demo which cost the most. (This has the effect of getting the salesperson's attention also...). Then compare with the rather cheaper pairs that you were actually thinking of. Having a point of reference at the time of listening is vital. Otherwise, things can get very, very subjective.
So. What were we actually talking about?
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Picked up the BTA-1 Bluetooth adapter for the F3, just messed around with it a bit, but seems real nice. Very straightforward and easy. Will report back on the range...
Edit: also, just can't say enough about the size of this thing (not the greatest pic, but you get the idea). Worked up a mount for it + the iphone i use for DPA recording. Very stoked about this setup to run 2 sources!
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Picked up the BTA-1 Bluetooth adapter for the F3, just messed around with it a bit, but seems real nice. Very straightforward and easy. Will report back on the range...
Edit: also, just can't say enough about the size of this thing (not the greatest pic, but you get the idea). Worked up a mount for it + the iphone i use for DPA recording. Very stoked about this setup to run 2 sources!
What did you use for the mounts here? Looks great!!
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Picked up the BTA-1 Bluetooth adapter for the F3, just messed around with it a bit, but seems real nice. Very straightforward and easy. Will report back on the range...
Edit: also, just can't say enough about the size of this thing (not the greatest pic, but you get the idea). Worked up a mount for it + the iphone i use for DPA recording. Very stoked about this setup to run 2 sources!
What did you use for the mounts here? Looks great!!
Thanks!
Phone mount: https://www.amazon.com/gp/product/B01M7RE39O/ref=ppx_yo_dt_b_asin_title_o00_s00?ie=UTF8&psc=1
Zoom mount: https://www.bhphotovideo.com/c/product/831804-REG/Impact_me_100_Atom_Clamp_with_Ratchet.html
I had the impact clamp already, and just found a spigot that fit the Zoom. Could probably improve that one a bit, but it's solid as-is.
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Running a battery life test and messing around with this for the first time. I'm running Berliner CM33 mics directly in via XLR - a bit confused by the setting options. For each input you can choose:
Line in
Line in +24v
Line in +48v
Mic In
Mic In +24v
Mic In +48v
Since I'm not running through a battery or pre-amp, I do need phantom power. I assume mic in +48v is what I need?
Been a while since I've had to think about this question...thanks!
Also, in terms of magnification, what has everyone been using?
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Running a battery life test and messing around with this for the first time. I'm running Berliner CM33 mics directly in via XLR - a bit confused by the setting options. For each input you can choose:
Line in
Line in +24v
Line in +48v
Mic In
Mic In +24v
Mic In +48v
Since I'm not running through a battery or pre-amp, I do need phantom power. I assume mic in +48v is what I need?
Been a while since I've had to think about this question...thanks!
Also, in terms of magnification, what has everyone been using?
Yes, mic in +48. Line in with +24 or +48 V phantom is used when you have mics with a super high output level.
I have an F6; not and F3 but I think the magnification/level doesn't matter. Just adjust in post.
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So I'm trying to wrap my head around the magnification/level function. Is it there to make the user think they have some sort of gain control, but it has no effect on the end result?
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So I'm trying to wrap my head around the magnification/level function. Is it there to make the user think they have some sort of gain control, but it has no effect on the end result?
It has an effect on the end result only when it is set before the recording is started. Adjusting after recording has started has no effect on the final WAV.
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I think the strategy with the magnification level is to use it chiefly to ensure that when incoming levels are low, you still can see that all is well (that you are actually getting a signal) from the movement of the display. It's likely then to appear to clip at high incoming levels, but that is fixable in post production. The digital level is set by the magnification value existing when the recording starts. You can then change it, but that won't affect the original value being applied to the recording (which of course would be counter productive). What it will do is to affect the monitoring level in headphones. So don't raise the magnification level radically if the existing level in your headphones is loud enough - mind your hearing! If you do want to raise the magnification level significantly during recording, first reduce the headphone level a fair bit, and readjust it after changing the magnification level.
Note that I don't actually have one... but I think my advice is correct.
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I don't have one either.. and believe your understanding to be correct.
One qualification on this part-
It's likely then to appear to clip at high incoming levels, but that is fixable in post production.
This is true for for low input levels as discussed. It's also true for higher input levels as long as the input is not so hot as to clip or otherwise overdrive the analog input stage of the recorder ahead of it's 32-bit float ADC stage -- hence the provision of both mic and line input sensitivities.
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Can anyone tell me how these would show up on metal detector wands and walkthroughs?
Also interested about the same with mixpre
Cheers
Duncan
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Its metal...it will show up
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Can anyone tell me how these would show up on metal detector wands and walkthroughs?
Also interested about the same with mixpre
Cheers
Duncan
Tell 'em it a Rubik cube
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"It's my cigarette case. Very small cigarettes are much better for my health."
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I just got mine today. Some thoughts: the battery indicator does not always show up. Not sure why this is. You can power it through the USB-C, and this did work plugging it into an AC adapter. I had the thing running on some Powerex fully charged AAs for a few hours (didn't time it but I was working for several hours, it was running at 48V phantom). So certainly you'd want a box of AAs in a festival setting or an external battery pack.
I was hoping it would be a better stealth unit. The heavy metal frame seems to be possibly removable, and would leave you with a VERY small and VERY light unit that would be pretty much the be all and end all of all in tiny phantom/pre/recorder units. I did try removing the two long mount screws to see if the metal part would slide off (it looks like it would and should), but it didn't, at least with an amount of force I'd be comfortable with on a brand new piece of gear. There may be more screws under the mounts for the belt loop attachments but again, not comfortable prying these off either. A search of Youtube and Google didn't turn anything up but hopefully someone will work out how we can set this metal frame aside for stealth missions.
I see in the review someone posted that the external bluetooth app would definitely be something you want to augment the tiny and bare bones display on this.
Edit: I got a little farther with disassembly without completely screwing it up, there are also 2 small screws under the battery, and the mic input screws need to be removed. At that point you need to slide the back cover past one of the battery compartment springs which isn't easy, then there also seems to be some stuff connected inside that back panel that will be connected by folded cable bundles. Perhaps the next step would be getting that back panel off, then there would be mount screws to ditch the metal part of the chassis. My dad used to hate it when I took apart his stereo, and I haven't learned too much since then.
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I just got mine today. Some thoughts: the battery indicator does not always show up. Not sure why this is. You can power it through the USB-C, and this did work plugging it into an AC adapter. I had the thing running on some Powerex fully charged AAs for a few hours (didn't time it but I was working for several hours, it was running at 48V phantom). So certainly you'd want a box of AAs in a festival setting or an external battery pack.
I was hoping it would be a better stealth unit. The heavy metal frame seems to be possibly removable, and would leave you with a VERY small and VERY light unit that would be pretty much the be all and end all of all in tiny phantom/pre/recorder units. I did try removing the two long mount screws to see if the metal part would slide off (it looks like it would and should), but it didn't, at least with an amount of force I'd be comfortable with on a brand new piece of gear. There may be more screws under the mounts for the belt loop attachments but again, not comfortable prying these off either. A search of Youtube and Google didn't turn anything up but hopefully someone will work out how we can set this metal frame aside for stealth missions.
I see in the review someone posted that the external bluetooth app would definitely be something you want to augment the tiny and bare bones display on this.
I have the Bluetooth adapter but find no need for it. I am very familiar with 32bit recording having had the F6 and mixpre6II for some years now. The difference in concept is slow to be understood, but recording with these decks is not at all like 24bit or any other deck. I am sure people will disagree or not understand, but that is okay, I record a fair amount and am happy with what i have discovered.
On the F3 in particular, part of the reason for the wave settings showing versus meters is that there is no need for meters and really no need for a screen audio-wise. The concept that is most misunderstood is that you do not set the gain. It is pre-set by the machine and therefore meters are not relevant. This is a true set and forget deck. You do set the faders, or the output, and that is how loud or low your volume will be when you apply to post. But it is just the output volume, the faders, not the gain. So you can change your recording in post if too low or too loud and there is no loss of quality, no difference because it is just output (volume). Input (gain) was set by the deck.
I have come to realize this deck in particular is not like anything else. It is a game changer in approach to recording and especially great where there are unknowns at the start of a show, because it does not matter.
So for the first time when I use this deck, I shut the lighting on the screen off, hit record and am done. The recordings are stellar in terms of the deck. It is still up to me for placement etc., but the deck is flawless and requires no adjusting. I found the bluetooth part unnecessary.
I will say this. If you do not do any post work on your recordings, if you just somehow listen without using a post application, then this is the wrong deck for you. Then you will need to figure out the faders and volume (not the gain as you have no ability to adjust), so you may as well use a different deck, the advantages of this deck are largely nullified.
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dallman,
Looking forward to checking out your F3 the next time that I see you! Thanks for posting.
David
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Listening to my first pulls with the F3 from a festival from last weekend. A few thoughts:
-So far, I like what I hear in terms of sound. I've been running a Oade modified R44 in recent years, and by no means have I done any direct comparisons, but thus far I don't hear any thing that I don't like with the F3
-The bluetooth function using the phone is nice, but I don't like that if it gets disconnected it just cuts it off with (seemingly?) no way to reconnect. So if you walk too far away for whatever reason, just know that you won't be able to monitor things on your phone again without reconnecting on the device. Maybe no way around this, dunno.
-All of the recording was done at an outside festival, mics probably 80-100' back? Folk / Rock music, but the things basically peaked at around -12db, so didn't really get to run things especially hot. But nonetheless, happy with how things turned out.
-Have it rigged so I can clamp the device + a power source direct to the stand, and really loved not having my gear on the ground.
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It might be in here somewhere but does anyone know if the f3 has m/s decoding?
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No M/S decoding and monitoring. Dealbreaker for me, and a strange omission given the apparent target audience. And not hard to do.
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The Ron Popeil recorder, set it & forget it... 32 bit, auto levels, stereo with P48s.
I definitely see one in my bag this year.
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I had too many beers and ordered one off amazon with 70 bucks off from points..... Goddamnint. more stuff. be here in time for fool's paradise though. My thought was I could return it if I don't like it. :shrug:
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I knew nothing of 32-bit floating point before I learned of the F3. So basically you just plug in your mics and press record? And then in post what would you have to do? Adjust the volume, normalize, whatever you normally do when you're editing? What kind of audio file does the F3 store? Is it a wave or something else? Do the F6 and the new mixpre's behave the same? The mixpre-3 and 6 ii both look like they have traditional gain knobs.
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Short answer, yes. Pull the overheated 32 bit file into audacity and normalize it. I have an F6 and like the set it and forget deal especially with mics near the stage.
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Yes that's very tempting. I usually mess with my levels during the first song, unless the band allows me to set up during sound check. When I'm not seated near my gear, it would be useful to just press record before they actually take the stage and then walk away.
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I knew nothing of 32-bit floating point before I learned of the F3. So basically you just plug in your mics and press record? And then in post what would you have to do? Adjust the volume, normalize, whatever you normally do when you're editing? What kind of audio file does the F3 store? Is it a wave or something else? Do the F6 and the new mixpre's behave the same? The mixpre-3 and 6 ii both look like they have traditional gain knobs.
Mix Pre-3 v2. is going to be more versatile, have gain structure to tweak and likely be worth every penny, but I like to carry light and lower investment.
In in NYC, I go to sketchy watering holes sometimes, walk a few blocks back to street parking, and don't want to feel like I have to protect a bank vault of gear in my bag.
I'd love to hear a 2 mic A-B of an F3 and a Mix Pre 3, just for comp.
Things occuring in "the digital realm" have artifacts and unique character. ...But usually less so than an analog device
Still, those inputs to the F3 aren't digital, so...
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I'd love to hear a 2 mic A-B of an F3 and a Mix Pre 3, just for comp.
130-140dB and hot-out mic, that would be interesting.
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I'd love to hear a 2 mic A-B of an F3 and a Mix Pre 3, just for comp.
130-140dB and hot-out mic, that would be interesting.
I have been meaning to do one of these comparisons. Might make it happen soon. I also want to know if I am missing out on any performance/tonality when comparing the zoom pres to the mp3's
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I also want to know if I am missing out on any performance/tonality when comparing the zoom pres to the mp3's
I think the difference that can be heard may be in the preamps.
All other preferences will depend on your production process probably.
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Unless differences are really significant, it's actually very hard for most people to spot the differences between recordings made on different devices, assuming levels are matched and the same mics etc are used. In the good old days, when recording classical CDs I would sometimes use short sections from MiniDisc inserted into the DAT material. The reviewers never noticed anything amiss - they were usually complimentary about the sound overall. But to read some people's scathing opinions of MiniDisc sound, you would have thought it would stand out like a sore thumb. (I used MD as a backup medium, and its pre-record ability was sometimes very handy when something unexpected happened in the sessions).
Of course with these 32 bit float recorders, a comparison could be made of apparently over-cooked recordings in 24 bits and 32 bit float, and there the difference would be very evident - horrible digital distortion vs nicely normalised pristine audio.
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Unless differences are really significant, it's actually very hard for most people to spot the differences between recordings made on different devices, assuming levels are matched and the same mics etc are used. In the good old days, when recording classical CDs I would sometimes use short sections from MiniDisc inserted into the DAT material. The reviewers never noticed anything amiss - they were usually complimentary about the sound overall. But to read some people's scathing opinions of MiniDisc sound, you would have thought it would stand out like a sore thumb. (I used MD as a backup medium, and its pre-record ability was sometimes very handy when something unexpected happened in the sessions).
Of course with these 32 bit float recorders, a comparison could be made of apparently over-cooked recordings in 24 bits and 32 bit float, and there the difference would be very evident - horrible digital distortion vs nicely normalised pristine audio.
That's a fun story about MiniDisc. I knew people that were using it, but never tried myself. In the situations you described, were you using the Hi-MD PCM format, or the earlier lossy compressed format?
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I was using full sized machines (in that context - I also had small ones for concert work, using Hi-MD) and I think that Hi-MD wasn't available on those decks. Scary how I forget now, when I used that medium so much back in the day. I even made a jazz CD using an 8 track Yamaha MD machine - I just found sample tracks from that at https://nz.7digital.com/artist/mike-garrick-band-and-anita-wardell/release/down-on-your-knees-649197
Now what were we talking about? Ah yes, the Zoom F3... show me a topic and I'll drift it...
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Do I need to convert to 24bit for playback? Any other workflow items I should be considering?
First time w/ 32bit...so just realizing I should probably think through some of this prior to working on some of these recent recordings!
Thanks...
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Do I need to convert to 24bit for playback? Any other workflow items I should be considering?
First time w/ 32bit...so just realizing I should probably think through some of this prior to working on some of these recent recordings!
Thanks...
I use a Mixpre 6 ii, but I reduce the gain on the WAV files to under clipping and dither to 24 bit to save as a master
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I use a Mixpre 6 ii, but I reduce the gain on the WAV files to under clipping and dither to 24 bit to save as a master
Is it necessary to dither to 24bit? What is the benefit of doing that?
Interestingly re: levels/clipping, I've found levels to be a bit low thus far with the F3. Don't think it's a power issue, but I've had to boost levels so far. My understanding is that 32bit helps with that too, in terms of noise...but thats a bit out of my depth. Anyway, it sounds good.
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I use a Mixpre 6 ii, but I reduce the gain on the WAV files to under clipping and dither to 24 bit to save as a master
Is it necessary to dither to 24bit? What is the benefit of doing that?
Interestingly re: levels/clipping, I've found levels to be a bit low thus far with the F3. Don't think it's a power issue, but I've had to boost levels so far. My understanding is that 32bit helps with that too, in terms of noise...but thats a bit out of my depth. Anyway, it sounds good.
Here is what you may (or may not) realize if your output is low or high...
Although gain is set on the F3 and cannot be changed, the output from the F3 (the volume level transferred from your F3 recording to your computer or whatever) that will show up when you transfer the file will be determined by what you set on the front screen of your F3 (the wave size showing). If you adjust your volume to be high when recording (big wave shows on screen), your output will be high. If you set it low (small wave shows on screen), your output will be low. It is not all that relevant as the gain has been set and the final product will not suffer whether your volume was set low or high, but it might make your post adjustments easier or harder. You cannot set or change the gain, but you can set or change the output file which is the volume of the wave file being transferred.
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I just did a battery test with some fully charged Powerex AAs (2.3AH each) and got 3hrs 45min of recording time, low volume stereo. Powering Telefunken M62 at 48V (you can set to 24V as well, haven't tried that yet- any comments there?) I'm pretty sure the phantom sucks more juice with high SPL shows but that's a pretty good baseline. I'd probably be able to get all of Phil but I'll swap out at set break and feel pretty safe with that. I use the Powerex charger which allows you to cycle the batteries at a programmable rate and number of cycles. I've found that a single cycle doesn't get you the full potential of the battery but takes up to 24 hours. Exited gracefully saving the files including the short one. Also just did a hot swap test starting with AAs, plugging in a little phone charger battery, pulling the AAs, then putting the AAs back in, pulling the charger all without losing recording. You don't have to hide a small phone charging battery, toss it in the tray going in.
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I just did a battery test with some fully charged Powerex AAs (2.3AH each) and got 3hrs 45min of recording time, low volume stereo. Powering Telefunken M62 at 48V (you can set to 24V as well, haven't tried that yet- any comments there?) I'm pretty sure the phantom sucks more juice with high SPL shows but that's a pretty good baseline. I'd probably be able to get all of Phil but I'll swap out at set break and feel pretty safe with that. I use the Powerex charger which allows you to cycle the batteries at a programmable rate and number of cycles. I've found that a single cycle doesn't get you the full potential of the battery but takes up to 24 hours.
I opted to just go with an external power source, but when I did some testing w/ Amazon & eneloop rechargeables I got about the same when supplying phantom power.
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I use a Mixpre 6 ii, but I reduce the gain on the WAV files to under clipping and dither to 24 bit to save as a master
Is it necessary to dither to 24bit? What is the benefit of doing that?
Interestingly re: levels/clipping, I've found levels to be a bit low thus far with the F3. Don't think it's a power issue, but I've had to boost levels so far. My understanding is that 32bit helps with that too, in terms of noise...but thats a bit out of my depth. Anyway, it sounds good.
Here is what you may (or may not) realize if your output is low or high...
Although gain is set on the F3 and cannot be changed, the output from the F3 (the volume level transferred from your F3 recording to your computer or whatever) that will show up when you transfer the file will be determined by what you set on the front screen of your F3 (the wave size showing). If you adjust your volume to be high when recording (big wave shows on screen), your output will be high. If you set it low (small wave shows on screen), your output will be low. It is not all that relevant as the gain has been set and the final product will not suffer whether your volume was set low or high, but it might make your post adjustments easier or harder. You cannot set or change the gain, but you can set or change the output file which is the volume of the wave file being transferred.
Lol, so I read through this thread several times & thought I understood that part of it, but clearly I didn’t!
Assuming the mics or setup isn’t being physically overloaded in any way, do we know if it’s better to have a BIG wave form that shows clipping (requiring lowering it in post) or have it say -24 and have to amplify in post?
Traditionally I know you don’t want to add any more gain than you have to, as you’re raising the volume on a lot of noise too…is that still the case w/32bit?
Anyway, thank you for finally helping me understand this!
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I use a Mixpre 6 ii, but I reduce the gain on the WAV files to under clipping and dither to 24 bit to save as a master
Is it necessary to dither to 24bit? What is the benefit of doing that?
Interestingly re: levels/clipping, I've found levels to be a bit low thus far with the F3. Don't think it's a power issue, but I've had to boost levels so far. My understanding is that 32bit helps with that too, in terms of noise...but thats a bit out of my depth. Anyway, it sounds good.
Here is what you may (or may not) realize if your output is low or high...
Although gain is set on the F3 and cannot be changed, the output from the F3 (the volume level transferred from your F3 recording to your computer or whatever) that will show up when you transfer the file will be determined by what you set on the front screen of your F3 (the wave size showing). If you adjust your volume to be high when recording (big wave shows on screen), your output will be high. If you set it low (small wave shows on screen), your output will be low. It is not all that relevant as the gain has been set and the final product will not suffer whether your volume was set low or high, but it might make your post adjustments easier or harder. You cannot set or change the gain, but you can set or change the output file which is the volume of the wave file being transferred.
Lol, so I read through this thread several times & thought I understood that part of it, but clearly I didn’t!
Assuming the mics or setup isn’t being physically overloaded in any way, do we know if it’s better to have a BIG wave form that shows clipping (requiring lowering it in post) or have it say -24 and have to amplify in post?
Traditionally I know you don’t want to add any more gain than you have to, as you’re raising the volume on a lot of noise too…is that still the case w/32bit?
Anyway, thank you for finally helping me understand this!
It is tricky and simple, a different learning curve than we are used to.
As far as setting the wave file on the front of the F3, as it is just output volume, it does not matter how you set it, there is no difference, or to put it more simply, nothing you do is affecting the actual recording. I'd say personal preference. Whether the wave file is tiny or large, the recording gain is exactly the same. It is only the output which is after the recording , that is being affected by your settings. I like to set my F3 medium to larger, but it really is just personal preference and maybe with some post programs (I use sound forge) it may be easier reducing or enlarging.
As the gain is pre-set by the recorder, you are not introducing noise if you record with your output (the wave file you set then see on the front of the F3) looking very small.
I hope this helps.
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At risk of being shot down by my elders and betters, I think this very recent article explains things very well.
https://www.wired.com/story/32-bit-float-audio-explained/
In my own head - FWIW - I think of it like this. Imagine you are looking at a track on a DAW, in other words, a graphical representation of the sound which looks like a kind of ribbon. In 16 bit or 24 bit audio, that track or ribbon isn't very wide. If your recording gain in the analog domain is wrongly set, your audio could go off the top of the ribbon, or off the bottom of it, losing data and/or causing distortion on the way. But with 32 bit float, the track or ribbon is so wide that your audio is like a stripe running in the middle of the length of the ribbon. The height of that audio stripe represents the difference between the lowest output of the analog part of the recording system (mic and preamp usually) and the highest possible output of that analog part. But the height of it is very small compared with the height of the 32 bit float ribbon. It doesn't matter where in relation to that ribbon it sits. You'll never get the audio ribbon to get anywhere near the top of the 32 bit float ribbon, nor near the bottom. You can set the "magnification" in the case of the F3 such that the audio sits a bit above the middle or a bit below, but it doesn't matter where. When you process it in your DAW later, you will be feeding it into something akin to a 24 bit or 16 bit bottleneck - and you can at that point shift the stripe of audio data up a bit or down a bit on the overall 32 bit float ribbon so that it goes into the bottleneck at an optimum height.
I now await others to say "Actually, that's crap" or "that makes sense" or "I didn't understand a word of that and fell asleep reading it." Time will tell. Gulp.
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back to the battery life for a second. I got mine yesterday, and had a show to record last night, I used it as a second recorder luckily because the band needed the tracks for a video shoot/promo they are working on. I unboxed it, messed with it long enough to set it up, no more than an hour I don't think. Did a little test recoding for 30 minutes. Then I took it to the show, and it did not make it through the first set, which really suprised me. I had a battery swap ready for the second set. External will be ideal for this, or at worst a battery change before every set. Might be my rig? Akg CK3X>PFA's?
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back to the battery life for a second. I got mine yesterday, and had a show to record last night, I used it as a second recorder luckily because the band needed the tracks for a video shoot/promo they are working on. I unboxed it, messed with it long enough to set it up, no more than an hour I don't think. Did a little test recoding for 30 minutes. Then I took it to the show, and it did not make it through the first set, which really suprised me. I had a battery swap ready for the second set. External will be ideal for this, or at worst a battery change before every set. Might be my rig? Akg CK3X>PFA's?
^
What batteries were you using?
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Perhaps the PFAs are also a power draw, being a preamp of sorts? I still like the idea of having backup AA and being able to hot swap using a small phone charger instead of a big battery pack.
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I’d be interested in seeing some tests with small external batteries to see how long you could get.
My maiden voyage was a festival where I basically set it and left it all day, so I just snagged a mid sized anker pack and it would be enough to run for days. But I’m sure there’s a sweet spot with smaller batteries too.
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At risk of being shot down by my elders and betters, I think this very recent article explains things very well.
https://www.wired.com/story/32-bit-float-audio-explained/
In my own head - FWIW - I think of it like this. Imagine you are looking at a track on a DAW, in other words, a graphical representation of the sound which looks like a kind of ribbon. In 16 bit or 24 bit audio, that track or ribbon isn't very wide. If your recording gain in the analog domain is wrongly set, your audio could go off the top of the ribbon, or off the bottom of it, losing data and/or causing distortion on the way. But with 32 bit float, the track or ribbon is so wide that your audio is like a stripe running in the middle of the length of the ribbon. The height of that audio stripe represents the difference between the lowest output of the analog part of the recording system (mic and preamp usually) and the highest possible output of that analog part. But the height of it is very small compared with the height of the 32 bit float ribbon. It doesn't matter where in relation to that ribbon it sits. You'll never get the audio ribbon to get anywhere near the top of the 32 bit float ribbon, nor near the bottom. You can set the "magnification" in the case of the F3 such that the audio sits a bit above the middle or a bit below, but it doesn't matter where. When you process it in your DAW later, you will be feeding it into something akin to a 24 bit or 16 bit bottleneck - and you can at that point shift the stripe of audio data up a bit or down a bit on the overall 32 bit float ribbon so that it goes into the bottleneck at an optimum height.
I now await others to say "Actually, that's crap" or "that makes sense" or "I didn't understand a word of that and fell asleep reading it." Time will tell. Gulp.
This is a very good explanation of the practical result of recording on a Zoom F-series or Sound Devices MixPre-II recorder. One significant thing missing both from your explanation and that Wired article is that you can only get these recordings to be what they are in the first place if you use multiple ADCs controlled by a DSP system which routes signals of different levels to different ADCs depending on their level before they are dumped in a 32-bit float bucket. Zoom uses two ADCs; Sound Devices use three, and I don't how many StageTec uses for their setup which has existed since the late '90s.
If you didn't have such a setup then you still would have the problem of very low levels getting buried in the noisefloor and/or very high signals clipping the ADC, no matter how enormous of a "bucket" a 32-bit float file gives you. I think of it like having both a small trowel and a full-size backhoe on site for a material sorting job that classifies rocks over a wide range of sizes. If you only had one or the other, or just something in the middle like a Bobcat, you would miss material at both size extremes and your huge sorting table with all its different size trays would be unnecessary.
///
Of course, the end user doesn't necessarily need to know any of that to get all the benefits of these multi-ADC 32FP systems; they "just work", and do so seamlessly. I do, however, think it is helpful to know that recording in 32-bit float on its own using a single ADC does not allow you this glorious freedom from setting levels. Case in point: I have a MOTU M2 interface for my PC, and it will output 32-bit float format to any software that supports it. It does not have auto-ranging multiple ADCs though, so while the 32-bit float recordings I track in Reaper give me a little bit of wiggle room, I still need to set input gain on the M2 just as carefully as I would when recording in 24-bit fixed.
The last point to remind everyone of is one that Gutbucket and a few others regularly remind us about: Even when recording with one of these new Zoom or SD recorders, you still have to make sure you are not clipping your mic/line input in the analog domain before it hits the fancy multi-ADC. If you record very loud things and/or use very sensitive mics with hot outputs, you may need to use mic pads or select less sensitive mics. Pay attention to "max input level" on spec sheets. SD MixPre-II can take much higher mic input levels than Zoom F series, but Zoom has a LINE+PHANTOM input setting you can select which allows you to record with 48V mics at LINE level, and that raises the max input level up to be comparable to the MixPre-II.
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The last point to remind everyone of is one that Gutbucket and a few others regularly remind us about: Even when recording with one of these new Zoom or SD recorders, you still have to make sure you are not clipping your mic/line input in the analog domain before it hits the fancy multi-ADC. If you record very loud things and/or use very sensitive mics with hot outputs, you may need to use mic pads or select less sensitive mics. Pay attention to "max input level" on spec sheets. SD MixPre-II can take much higher mic input levels than Zoom F series, but Zoom has a LINE+PHANTOM input setting you can select which allows you to record with 48V mics at LINE level, and that raises the max input level up to be comparable to the MixPre-II.
Good explanation with magnetic tape.
MP3ii has the ability to adjust the level of the microphone channel. I roughly know what level the performers will have or what I will record. For a long time I did not manage to feel all the joy from the 32FP. But a few times 32FP saved me with unfamiliar music in an unfamiliar place when I set the pianissimo gain level too high. There is no doubt that 32FP is useful for recording unpredictable events like thunder, industrial noises or gunshots.
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There is no doubt that 32FP is useful for recording unpredictable events like thunder, industrial noises or gunshots.
It is also a game-changer for those of us who are both performer and recording engineer, which is more than half of everything I record.
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I absolutely enjoy this discussion.
Let's consider the old feature of auto-levels, ALC, where a circuit dynamically adjusted input gain to a median and circuit-built top limit.
Are the F6 and F3 making any analog input adjustments to normalize levels into the ADC ?
Is it possible to top the analog limits on the input circuit, and what are the artifacts?
Is it "splattery", "harsh", "choppy", blank or lowered in level?
Similarly, what happens at the very low end, when sounds barely break above a mic's internal noise ?
Having dynamic range beyond all hearing is great, but when it is adjusted to within the audible range, what do we actually get from the theory and grand maths?
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I would summarise it in this way - 32 bit float doesn't overcome the limitations of the analog side of your rig (mics and preamps and mixers) but it takes nothing away. It's kind of like having the mic plugged into your ears, you could say. But perhaps it's best to think mainly in terms of the sheer convenience of doing away with level setting. That said, I used to delight in recording a symphony concert, and getting to the end with the peak level of the recorder showing I'd got to -1dB of full scale - and not over. Simply aiming for -12dB or -6dB was no way to pass the evening away... My theory that there is (in acoustic music) only one level made it not too hard to get it right. (Mic goes close to quiet guitar, further away from a loud orchestra, SPL at the capsule ends up much the same). Adding that theory to the theoretical dynamic range of a 32 bit float recorder means that it all becomes pretty low stress. If I had one of these devices I would set the magnification to show that the mics were working even with empty hall ambience (for total reassurance) and probably I wouldn't look at the display again.
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In my own head - FWIW - I think of it like this. Imagine you are looking at a track on a DAW, in other words, a graphical representation of the sound which looks like a kind of ribbon. In 16 bit or 24 bit audio, that track or ribbon isn't very wide. If your recording gain in the analog domain is wrongly set, your audio could go off the top of the ribbon, or off the bottom of it, losing data and/or causing distortion on the way. But with 32 bit float, the track or ribbon is so wide that your audio is like a stripe running in the middle of the length of the ribbon. The height of that audio stripe represents the difference between the lowest output of the analog part of the recording system (mic and preamp usually) and the highest possible output of that analog part. But the height of it is very small compared with the height of the 32 bit float ribbon. It doesn't matter where in relation to that ribbon it sits. You'll never get the audio ribbon to get anywhere near the top of the 32 bit float ribbon, nor near the bottom. You can set the "magnification" in the case of the F3 such that the audio sits a bit above the middle or a bit below, but it doesn't matter where. When you process it in your DAW later, you will be feeding it into something akin to a 24 bit or 16 bit bottleneck - and you can at that point shift the stripe of audio data up a bit or down a bit on the overall 32 bit float ribbon so that it goes into the bottleneck at an optimum height.
Extending this analogy- as Ozpeter describes above, imagine the relatively narrow track ribbon width of 16 bit, a wider ribbon for 24 bit, and a much, much, much wider ribbon for 32-bit float, This represents the dynamic range limit of the file format used to contain the data. Okay, here's the important next part- There are three other dynamic ranges that represent other "ribbon widths" that need to be considered ahead of the file format. The first is the dynamic range of the signal we're recording. The second is the dynamic range of the analog signal path of microphones up to the analog to digital converter (ADC). The third is the dynamic range of the ADC. Each represents a separate domain that needs to fit within the range of each subsequent domain that follows.
Generally..
The dynamic range of the signal being recorded is smallest.
The dynamic range of the analog signal path is somewhat larger (this determines noise floor and overload level).
The dynamic range of the ADC may or may not be larger (but ideally should be).
The dynamic range of the file format may or may not be larger still (but ideally should be).
It's all about strategies to make sure the dynamic range of each domain is larger than the one which proceeded it, ideally setup such that no adjustments are required to make sure each will fit.
32-bit float recorders incorporate two new strategies to do this:
- The first is paralleled ADC's which incrementally increases the dynamic range of the ADC.
- The second is the 32-bit float file format which dramatically increases the dynamic range of the storage format.
Let's consider the old feature of auto-levels, ALC, where a circuit dynamically adjusted input gain to a median and circuit-built top limit.
Are the F6 and F3 making any analog input adjustments to normalize levels into the ADC ?
Astute observation, and essentially what the paralleled ADC's are doing, setup to do so in an automatic way. This has pretty much been determined to be audibly transparent.
The increases in dynamic range of the auto-ranging ADC and of the file format are sufficient to shift the "dynamic range bottleneck" to the domain of the analog signal chain feeding the ADC. The bottom of that range is the noise floor, likely determined by the self-noise of the microphones. The top of that range is determined by the overload point, or level of acceptable distortion prior to outright clipping, of something in that analog chain. Could be anything from the microphones to the preamps to the analog input stage of the recorder itself.
With regards to the 32-bit recorder part alone, it's maximum dynamic range limits are now determined by the limits of it's analog input stage, not the ADC or file format.
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Is it necessary to dither to 24bit? What is the benefit of doing that?
Its always best practice to dither whenever reducing bit length. There isn't really any reason not to other than eliminating it as an extra step.
Is it necessary? Probably not. If the least significant bits being truncated are well below the noise-floor, the random noise of the noise-floor itself essentially acts as dither. Sound Devices doesn't bother to dither internally when storing a 24 bit file rather than a 32-bit float file on the recorder. They do dither if set to store a 16-bit file.
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Just got back from 3 days at a festival. I recorded for 8-9 hours per day using pfa and schoeps capsules at +48v phantom. A pair of eveready lithium batteries lasted all day each of the days. I went through 3 pairs in 3 days. That included some playback after the shows.
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I want to start off by saying I think the technology that allows 32 bit float to operate in a portable recorder is amazing.
That being said, to me the benefits for the actual recording are dubious and this point has been made numerous times in this thread.
Let's start with these facts.
1. The maximum dynamic range of the best analog signal capture/amplification equipment made generally stops at 120db.
2. Rock concerts have maximum sound pressures coming from the stage/PA between 90db to 110db on average, depending on the venue.
3. Typical concert crowd ambiance averages 40db (for between songs babble and shuffling around, not cheering/roaring applause)
4. The dynamic range potential of the 24bit PCM format is 144db.
So, if you were to record a rock concert soundcheck in an otherwise pin drop quiet empty room the dynamic range could theoretically reach 120db.
If using a 24 bit A/D with levels peaking at -12db, the analog noise floor will be down 120db below that point, or -132db, with 2 bits to spare.
When the audience comes in and fills up the seats, the effective dynamic range of the music falls to 80db, since the crowd chatter and racket becomes the new noise floor at -40db.
Since -60db to -50db is the lowest level mark on the meters of most recorders, even a recording that only occasionally lights up the -40db mark and is gain boosted in post will still retain 104db of dynamic range.
Bottom line, with the same analog signal source it's not possible for a recording in 32bit float to exceed the practical dynamic range of a 24 bit recording, even with levels barely hitting -12db on the meters.
So, what's the big draw of 32 bit float in a field recorder? I see it as primarily to benefit foley, ENG, and nature recording challenges. If I was recording a frog sitting on a rock in a stream croaking, and BOOM, a crack of thunder happens, I'd feel far more confident setting levels with 32 bit float recorder than 24 bit recorder.
For concert recording, I just don't see 32 bit float being essential or substantive to proper level setting.
With all that said, if I was in the market for a 2 channel recorder I could use right out-of-the-box, the F3 is at the top of my list, but not because it's format is 32 bit float.
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^ To me, in the context of recorders with two (or more) 24-bit ADCs of differing gains, I think the big advantage of 32-bit float would be if you are doing mixing (post-ADC) on the recorder (where the combination of sources might exceed 0 dBFS). I suspect that there is a gain setting (maybe no gain) that would give an identical result in either 24-bit or 32-bit float for the individual channels. Maybe I'm wrong; the SD patent was pretty heavy reading!
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Datbrad- At the risk of going OT in the F3 thread. What you describe is essentially how I currently run F8 in 24 bit. I do tweak my recording levels, not because I need to in order to maximize dynamic range or avoid overload, but only to get the relative level between channels about where I want them for purposes of playback on the recorder..
I think the big advantage of 32-bit float would be if you are doing mixing (post-ADC) on the recorder (where the combination of sources might exceed 0 dBFS).
This is a good point. The only time I (rarely) have a problem with what I've recorded to F8 in 24bit mode is playback mixing on the recorder of recordings made of high SPL material, and then only because of the annoying quirk of Zoom's Mid/Side decoding implementation being pre-fader only. Basically, that implementation eliminates the ability to adjust mix level of Mid/Side decoded channels upon playback, so all other channels must be mixed around the level of those channels as recorded. If the omni channels then need a lot more fader level in the mix for instance, they can sometimes clip the mixer. That doesn't always happen, but it can with very loud material.
I suspect that there is a gain setting (maybe no gain) that would give an identical result in either 24-bit or 32-bit float for the individual channels.
I think you are correct. And it should if Zooms dual ADC auto ranging technique provides a wider dynamic range than that of the analog input front-end of the recorder.
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I think the whole thing can be viewed in reverse - if 32 bit float had been the norm for the last 50 years, and someone brought out a 24 bit recorder and said one of the features was that you can set the input level so that it doesn't overload or undercook, would people say, wow, that's a great idea?
Gutbucket, I'm not quite sure of the context but personally I was a heavy user of Voxengo MSED VST when remixing stuff in a DAW - that enables you to fiddle with mid / side levels to your heart's content with any stereo track, and of course is most appropriate if the source of the stereo track was an MS pair. It basically re encodes the signal back to MS, then allows you to change the MS balance, then decodes back to XY.
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Thanks Ozpeter- I also like and recommend Voxengo MSED VST. Its a great M/S tool, and does exactly what is intended without extraneous features. Its ability to balance imaging by allowing one to pan the Mid and Side channels separately is especially useful - the effect of which is to apply a different M/S ratio to the Left verses Right channel output. That can be done via a mixer as well, but its easy and straight forward using the plugin.
To clarify, the issue I describe above occurs only when mixing on the recorder itself and needing to add digital gain which causes some internal levels to exceed 0dbFS. Its not a problem when mixing on the computer, where levels can bee freely adjusted for all channels, processing is typically taking place in a 32-bit float calculation space, and MSED or similar functionality built-in to many DAW programs is used.
I think the whole thing can be viewed in reverse - if 32 bit float had been the norm for the last 50 years, and someone brought out a 24 bit recorder and said one of the features was that you can set the input level so that it doesn't overload or undercook, would people say, wow, that's a great idea?
Except in that case they either wouldn't be overloading or overcooking input levels to begin with (and thus wouldn't perceive a problem that needed solving), or if they were, that solution wouldn't fix it (because the analog input or ADC is what would be overloading, not the file-storage format that follows.
Its all about the order of the sequential domains and where overload is occurring, or is capable of occurring, based on the range of what preceded it: signal>analog-input-stage>ADC>file-format
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Thanks Ozpeter- I also like and recommend Voxengo MSED VST. Its a great M/S tool, and does exactly what is intended without extraneous features. Its ability to balance imaging by allowing one to pan the Mid and Side channels separately is especially useful - the effect of which is to apply a different M/S ratio to the Left verses Right channel output. That can be done via a mixer as well, but its easy and straight forward using the plugin.
To clarify, the issue I describe above occurs only when mixing on the recorder itself and needing to add digital gain which causes some internal levels to exceed 0dbFS. Its not a problem when mixing on the computer, where levels can bee freely adjusted for all channels, processing is typically taking place in a 32-bit float calculation space, and MSED or similar functionality built-in to many DAW programs is used.
I think the whole thing can be viewed in reverse - if 32 bit float had been the norm for the last 50 years, and someone brought out a 24 bit recorder and said one of the features was that you can set the input level so that it doesn't overload or undercook, would people say, wow, that's a great idea?
Except in that case they either wouldn't be overloading or overcooking input levels to begin with (and thus wouldn't perceive a problem that needed solving), or if they were, that solution wouldn't fix it (because the analog input or ADC is what would be overloading, not the file-storage format that follows.
Its all about the order of the sequential domains and where overload is occurring, or is capable of occurring, based on the range of what preceded it: signal>analog-input-stage>ADC>file-format
No matter how Old School one wants to be, I know some people like to set levels, and I know people who don't. If running right in front of the stage it is very nice to just set it and forget it, and not bother any other crowd or band members while checking in on things. The possible best feature of 32 bit float is noticed in post IMO. You take an "overdriven/clipping/float) file into your DAW that has NOT one volume adjustment up or down during the entire recording, so long as you leave it alone, and normalize it. What you end up with is a nice linear recording with no level ups and downs to deal with in post. It just reflects what the PA or other source was doing at the time. I feel that the F6 and F3 have made me a better taper and is a better experience in the field and in post. Ozpeter hit the nail on the head, at least for me. This is amazing technology, why not use it.
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The possible best feature of 32 bit float is noticed in post IMO. You take an "overdriven/clipping/float) file into your DAW that has NOT one volume adjustment up or down during the entire recording, so long as you leave it alone, and normalize it. What you end up with is a nice linear recording with no level ups and downs to deal with in post. It just reflects what the PA or other source was doing at the time.
Yes. I think Datbrad's point is that you can set things up so as to work that way already. Advantage is not needing to do any initial setting up.
I feel that the F6 and F3 have made me a better taper and is a better experience in the field and in post. Ozpeter hit the nail on the head, at least for me. This is amazing technology, why not use it.
Agreed. I also agree with Ozpeter's take on practical usage, effectively using it as a form of meter range adjustment.
If I had one of these devices I would set the magnification to show that the mics were working even with empty hall ambience (for total reassurance) and probably I wouldn't look at the display again.
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I have not used a 32 bit float device.
I find it pretty amazing that a person can more or less pull a great recording every time without worry of levels.
And I know it would be great in specific situations.
But… does this technology make a person a better taper??.. or does it make a person a worse taper that is able to make better tapes??
I’m sure I’ll get a F3 at some point.. so my question is just stirring the pot a bit.
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I have not used a 32 bit float device.
I find it pretty amazing that a person can more or less pull a great recording every time without worry of levels.
And I know it would be great in specific situations.
But… does this technology make a person a better taper??.. or does it make a person a worse taper that is able to make better tapes??
I’m sure I’ll get a F3 at some point.. so my question is just stirring the pot a bit.
That's my feeling as well. I want to see one and play with it before I would buy. Its more about ease of getting in and improving the recordings rather than making it easier.
Its like GPS and traveling. I remember doing Europe Dead tour by car using maps to navigate versus now my phone tells me where to go. Not sure I could navigate that trip uses maps today 30 years later.
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I have not used a 32 bit float device.
I find it pretty amazing that a person can more or less pull a great recording every time without worry of levels.
And I know it would be great in specific situations.
But… does this technology make a person a better taper??.. or does it make a person a worse taper that is able to make better tapes??
I’m sure I’ll get a F3 at some point.. so my question is just stirring the pot a bit.
OK to stir the pot... When I wrote that I meant to mention that it makes it easier to concentrate on other aspects of recording, like microphone choice, stand height, location,pattern etc. In the case of this past weekend when I did two nights at the same venue with the same 2 bands, I recorded from 3 different places in the amphitheater because it was not very sold, and things were loose. Had time to walk the room during the sets and see where sounded best in there... And at the end of it all, the levels become a non point of comparison. The setup mattered more.... I travel light, actives and 32 bit, so sure could I make a better recording with pre's/AD's and what not, probably. But I don't see myself doing that.
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In the case of this past weekend when I did two nights at the same venue with the same 2 bands, I recorded from 3 different places in the amphitheater because it was not very sold, and things were loose. Had time to walk the room during the sets and see where sounded best in there..
This and the decisions made based on it is what gets categorized in the category of important higher-level taper skill to my way of thinking. The other stuff all acts in support of the opportunities realized, and choices made based around this. Good to figure out ways of managing all the the rudimentary stuff so attention can be focused here instead so that good opportunities can be taken best advantage of. If 32 bit helps you to do that, then it's for the win.
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There is a similar movement going on in pro video. Lytro has come out with a 40K camera to capture what they call a "light field". Basically, with that resolution zooms and pans no longer have to be done live, all that can be done in post. You set the camera up and point it in the general direction of the subject and hit the rec button. The raw capture is just a fixed wide shot, no cinematography skills are required. Anyone with an index finger and enough aim to push a rec button can do it. Sound familiar?
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Audio recording is a combination of art and science. Microphone selection and placement is the art part. Everything downstream of the mics is the science part. It's the art part which distinguishes the great recordist from the lesser, and the great sound from the inferior.
(I guess dealing with Panatrope's second law of recording is also part of the art - "The number of channels required for a recording is always one more than the number of channels available".)
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I need two more!
Agreed, although a different flavor of art comes back into play in the mixing/mastering realms.
Sound familiar?
Visual ambisonics. Pick any pattern, pointed in any direction and derive as many virtual microphones as you want after the fact, but limited to that single point in space. If you want other camera angles you'll still need additional cameras / additional microphones if you want time-of arrival cues.
Sure would be cool shrunk down to a tiny Gopro-like format to mount on the mic stand, except for the crushing file size!
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Sure would be cool shrunk down to a tiny Gopro-like format to mount on the mic stand, except for the crushing file size!
GoPro Max 360. Records in 5k, lets you resolve down to a 1080p shot where you can adjust the POV pretty drastically in post. I use one kayaking. It's amazing. It's not the same as a good 4k cinema camera, but it's darn sure good enough for youtube.
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I have not used a 32 bit float device.
I find it pretty amazing that a person can more or less pull a great recording every time without worry of levels.
And I know it would be great in specific situations.
But… does this technology make a person a better taper??.. or does it make a person a worse taper that is able to make better tapes??
I’m sure I’ll get a F3 at some point.. so my question is just stirring the pot a bit.
I have been using the technology for some time now, and while it is all subjective, I'd say no, it has not made me a better taper. Having recorded steadily since 1971, I learn something every time out. It is mic placement, listening to the room or outdoors, adjusting for so many reason and I am a better taper today than I was in 1971. Actually I think it took until the late 1980's before any of my recordings were fit for public consumption. I love the tinkering, mixing, matching and the learning best of all about this hobby and passion. I do however really like the new technology and it is one less thing to worry about and it helps me and my process continue to evolve, as it has continually evolved since my first recording. That said, it still is experience that sets my recordings apart from my earlier ones, and allows me the opportunity to try to get the best recording in whatever set of circumstances I am handed each time I go out to record. There are always lots of little things done that affect the end result. Do I make better recordings in a venue I know inside out over one I am in for the first time? The odds are better I might as my experience is greater, but that does not make it a given. The entire beauty of why so many of us record is partly to have copies of shows, artists and music we love, but it evolves for so many of us that we love the actual experience of recording ourselves with each our own techniques and habits which we all believe will allow our recording to be the best one and our favorite version of all those that get made in a given evening.
Yug as you know as well as anyone who has done open recording with me I absolutely enjoy complicating the and regularly changing up my process, but that is just me doing what I love. ;D
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I have not used a 32 bit float device.
I find it pretty amazing that a person can more or less pull a great recording every time without worry of levels.
And I know it would be great in specific situations.
But… does this technology make a person a better taper??.. or does it make a person a worse taper that is able to make better tapes??
I’m sure I’ll get a F3 at some point.. so my question is just stirring the pot a bit.
Does tech that makes the process less complicated and less error-prone turn folks into worse tapers? Please help me understand that. Did longer tapes that made 2nd set flips easier make us worse?
Also, one thing I know for sure: anything that allows me to be more present in the moment to connect with the artists I'm seeing is a good thing. Quit taping for a bit because I felt I was getting too distracted from the music. Gear is fun, but art is where it's at for me.
YMMV 😀
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Damn those 80 minute Cd's. Useless. And FLAC you too! to the level of 8 haha! :cheers: beers are flowin...
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..like I said, just stirring the pot.
I understand the 32 bit float gives an opportunity to explore other techniques.
I was simply questioning if it makes a person a better taper??.
It seems to me that it takes away much of the effort (for good or bad) of recording a show.
All a person needs to do is hit record and enjoy the show.. which isn’t a bad thing.. but simplifies it to the point of being null and void somewhat it seems. Which will probably result in many more good sounding recordings and possibly turn on more people to becoming concert tapers.. which is great!
But I’m guessing that it’ll make tapers lose their ‘muscle’ a bit kinda like an electric bike does.
I have no problems with an electric bike.. it’s easier and a lot of fun I’m sure, but does it make you a better bicyclist??
Does a self-driving car make a person a better driver??
Does the 40K camera that datbrad mentioned make a person a better photographer??
Does Google make us smarter??
Do map books even exist anymore??
Do kids learn cursive these days, do people type on keyboards anymore (yeah yeah I’m plucking away with one finger on a touch pad, but that’s beyond the point). ;)
Is 32 bit float another step in dumbing down the human process??
Do humans learn without making mistakes??
IMO the 32 bit float can make a person a better post-production engineer.. which is a facet of releasing a good recording.. but is separate from the taping end of things IMO.
So will the F3 squash a Mytek or AD2K or SD or any other 24bit AD??
Or does it just make it easier??
That’s my question.
And is anybody running a preamp in front of the F3 or does that defeat the purpose?
Are all 32 bit float dual ADC’s the same?
It’s all about making great tapes and having fun while doing it..
Like I said, I’m sure that I’ll be ‘floating’ at some point.. I’m just slow, skeptical, and stubborn. :P
…yeah that was a bit overboard, but it’s Friday night!..
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Been thinking more about the application of gain in today's recorders and realized there's a distinct reason why 32bit float makes complete sense and leaving out traditional level controls a logical final step in the evolution of audio capture gear.
Most solid state recorders made today have trim knobs, but they are not true gain pots for adjusting preamp and/or analog gain ahead of the A/D. They are essentially for digital gain only. With fixed gain preamps and digitally controlled line levels in menu settings, having gain controls to adjust digital levels in the field isn't necessary.
Back in the day, we wanted to maximize the performance of preamps, since how hard they were driven made a difference in sound, and also prevent analog brick walling in the line level signal before it hits the A/D. With that structure now flipped, the only thing you can do wrong in the field is cause digital overloads. You can't stop analog brick walling, nor optimize the performance of the internal preamp like we did with recorders in the past.
Considering this, it occurs to me that 32bit float can't make you a better or worse taper, it simply removes a part of the equation that only has a negative risk, digital clipping, with no upside to being able to adjust that stage manually. With the evolution of recorders moving away from user control of any of the analog portion of the signal, if 32 bit float should become the standard for A/D, it's almost inevitable and probably should be.....
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^I get what your saying.. I think years ago jerryfreak on board here put together a rig that ran mikes straight into a 24 bit ADC. I think both for the reason that you said and for lower noise. I might be wrong on that but I think those were his reasons.
And if I ever said (which I don’t think I did) that 32 bit float makes a person a worse taper, I wasn’t being literal about it.
I was just stirring a ‘devil advocate’s’ attitude about someone stating that it makes person a better taper.
And my wacky rant in my previous post was continuing it.
The only thing that can make a taper a worse taper is not taping.
Apologies to all for derailing the thread.
It’s been quite a week and I was practicing some top notch ‘ass-hattery’. :smash:
If a recorder is in your pocket or gear bag, and the goal is to never look at it.. the F3 achieves that I think.
32 bit float aside, the F3 is exactly (minus the belt loops) what I’ve been waiting for.. a pocket sized recorder with full XLR’s, P48, and good enough battery life to run P48 for a full show.
Pretty amazing!!
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Esteemed Mr. North,
Your provocative points challenged many unquestioned assumptions. Thanks for that!
I own that I am a bit of a worrier, and that that can be distracting at a show. The combination of 9 hours of battery, 32 hours of SD card, and 32 bit float has been a game-changer for my experience. I imagine it's not so impactful for others.
One of the things that keeps me recording is that it continually presents opportunities to learn. Currently embarking on OMT4. Last time I ran more than 2 mics was Greeks '85, so really appreciating GutBucket's wisdom as a guide.
Don't see the need for any regrets. Your post was on topic and respectful as always.
Regards, Chris
[edited to fix typo]
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Currently embarking on OMT4. Last time I ran more than 2 mics was Greeks '85, so really appreciating GutBucket's wisdom as a guide.
Not to veer even more off topic (or to minimize Gutbucket's contributions to our little corner of the world which are considerable and much appreciated) but the stereophonic zoom paper was written by Michael Williams and published by the AES in 1984.
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3rd outing:
The F3 is continuing to be a winner. The gaff tape, while not ideal, is proving to be effective in avoiding erroneous stops. Battery life continues to be great with the Eneloop Pros - I am never below 50% by the end of a ~ 1.5 hour set @ 32/48 w/ 48v phantom on. Bag checks don't even see the Hermitshell case I have the F3 in opened up - it is small enough that it is entirely disregarded by security so far - obviously that will vary by venue. I'm going to do a Herbie show soonish - he tends to have much tighter screening from experience, so it will likely get its real test then.
In terms of ease of stealth operation, it really doesn't get any simpler. I am eyeballing levels based on house music before hitting record and dumping it till the end of show - not checking anything but an occasional glance at the red recording light (out of paranoia from the past lost recording on the first outing). Results have been great so far and the 32-bit would make even significantly botched levels a non-issue.
This time there was far too much bass so I've done some minor post to the samples as it was not listenable as it was. Still too much cymbal/hh "crashiness" going on for my taste so I did minor subtractive EQ on that as well.
All things considered, after 3 outings with different types of venues / different mic setups / etc I am giving this one a big thumbs up. As mentioned previously I am inclined to use the MixPre3-ii whenever there is no concern for security rummaging through the bag, based on my subjective assessment that it just sounds "better" raw from device, but when stealth is a factor I no longer feel like I am sacrificing as much as I did with the stealth oriented Sony A10, D50, D1, M10 / Sony XLR-1 / Battery Box setups I was using previously.
Recorded directly behind soundboard using a slightly narrow (practical stealth issues) NOS on stealth "object" peeking over the rail ~ 6ft high (very low "balcony") @ DFC, FOB with Line Audio CM4 > F3.
Artist: Keiko Matsui
Venue: Bethesda Blues and Jazz Club, Bethesda, MD.
Sample 1:
https://we.tl/t-Nc0HP9SPXl
Sample 2:
https://we.tl/t-kxPahRPcVZ
Follow on from the above - made it in to the Herbie show with a minor deflection. *Looks at device* ... "Is that recording equipment?" ... "No" ... "Thanks, keep your mask on during the show please!". Here's sample from the show with subtractive EQ to remove excessive bass and some added gain.
https://we.tl/t-pMfQN6tla4 (https://we.tl/t-pMfQN6tla4)
Line Audio CM4 > Zoom F3 (P48 on, 32 / 48) > r8brain dither to 24, 48 > FLAC.
Final verdict - I'm entirely sold on this thing. It has everything you could ask for (minus M/S decode) and nothing you wouldn't (e.g. sub-par built-in mics). As a bonus, it's discrete enough to pass through persnickety venue checks and can even dodge a minor security interrogation due to its size / appearance.
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Currently embarking on OMT4. Last time I ran more than 2 mics was Greeks '85, so really appreciating GutBucket's wisdom as a guide.
Not to veer even more off topic (or to minimize Gutbucket's contributions to our little corner of the world which are considerable and much appreciated) but the stereophonic zoom paper was written by Michael Williams and published by the AES in 1984.
Yes! I make no claims to the ownership of good ideas, but I will champion and downright proselytize for what has served to empirically illuminate and expedite my own pursuit of greater taper enlightenment. William's Stereo Zoom was and continues to be a foundational cornerstone in much of that. I believe the most tangibly useful single thing I've contributed to the taper community here is the improved PAS table - which at it's core is 100% William's Stereo Zoom. All I did was carefully think it through and repackage it in such a way as to make it directly applicable to tapers and easy to use.
However, I do think the OMT multichannel approach is different in an important way, being less derivative and representing something of a different paradigm entirely, even-though the basic Stereo Zoom relationship remains fundamental within it. The combination of multiple channels of a purposefully-arranged array of microphones set up so as to optimize the interactions between them represents something quite different, requiring the reassessment of what was widely known to work well for a pair of microphones alone. The basic acoustics and signal aspect underpinnings remain the same, but to effect something greater than what might otherwise come from the simple sum of stereo pairs requires the reconsideration of the ways in which a single stereo pair is best optimized. The parsing of what works for a single pair and also applies to the interactions of a purposefully designed multiple microphone arrangement verses what doesn't is something of a brave new world. My thanks to this forum as the place to discuss exploration of working this stuff out!
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So will the F3 squash a Mytek or AD2K or SD or any other 24bit AD??
Or does it just make it easier??
That’s my question.
I think it just makes it easier. 24-bit ADs are more than sufficient for any music recording (even 16-bit, for that matter).
And is anybody running a preamp in front of the F3 or does that defeat the purpose?
Are all 32 bit float dual ADC’s the same?
If you have a preamp with a flavor you like, or that pairs well with your mics, I think it still makes sense to run one. Also, as I understand it, there aren't really 32-bit float ADCs; these units use multiple 24-bit ADCs, with differing gains, and combine the results post-conversion.
With respect to Williams and the Stereophonic Zoom, I agree with Gutbucket. It's illuminating and time hasn't really dimmed it much. I recall reading that it was based on rather limited data, though, so maybe it could be tweaked a little. A couple of degrees or cm here or there, perhaps, but the take-homes probably wouldn't change much...
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The Williams curves are close enough for our use. There are a few alternate data sets that differ somewhat, I think partly based on the signal used as stimulus for the testing: impulsive clicks, sinewaves, noise, speech, music or whatever, but can't recall specifics as it's been years since I looked that far into it. [confirmed, see edit note below..]
For deriving all the values in the Improved PAS Table I used Eberhard Sengpiel's data, in the form of the online sengpielaudio visualizer (http://www.sengpielaudio.com/Visualization-ORTF-E.htm) rather than William's published curves.
Here are two links to PDF's hosted at www.sengpielaudio.com showing the differences in the underlying trade-off curves derived by various researchers:
http://www.sengpielaudio.com/InterchannelLevelDifferenceTimeDifference1.pdf (http://www.sengpielaudio.com/InterchannelLevelDifferenceTimeDifference1.pdf)
http://www.sengpielaudio.com/InterchannelLevelDifferenceTimeDifference2.pdf (http://www.sengpielaudio.com/InterchannelLevelDifferenceTimeDifference2.pdf)
Differences sure, but leveraging the general trend of the underlying relationship is what is important for informing microphone setup.
Edit-
Note found on the bottom of the page of the first PDF linked above reads:
Red curve: Master's Thesis by Gert Simonsen at the Technical University of Denmark Lyngby (1984), also
used by Michael Williams. Test signals: maracas and claves clicks. http://www.sengpielaudio.com/Claves.htm
Blue curve: Eberhard Sengpiel and tonmeister students (1992) used as test signals monaural human speech
and a Beethoven string quartet in a usual control room with Bowers & Wilkens 801 matrix 3 loudspeakers.
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^ Yes, that is the article to which I was referring. One thing I can say: I am very glad that Dr. Sengpiel's website lives on despite his passing. It might not be beautiful, but there is a TON of useful information there...
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FYI, the F3 looks like it is recording when it isn't. Make sure the red light is on. I thought I had a lot of time and told the GF how to start it, and apparently she hit record twice which turns it back off. https://archive.org/details/deadco2022-06-14.m60.flac16 (https://archive.org/details/deadco2022-06-14.m60.flac16) Second recording with my new Telefunken M60s, they are flat out amazing.
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Hope you weren't too hard on her, sounds like an easy mistake to make during daylight hours. I know it burns memory, but I usually start the deck if I need to step away within 30 min of the expected start time. If I get back in time, I just stop the recording and restart it to create a new file, then later I delete the junk file.
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back to off topic, Neumann has a great app called Neumann Recording Tools, for android and iOS, which calculates recording angle for nearly any pair.
review here: https://www.soundonsound.com/reviews/neumann-recording-tools
https://apps.apple.com/us/app/recording-tools/id576702914 (https://apps.apple.com/us/app/recording-tools/id576702914) iOS
https://play.google.com/store/apps/details?id=air.com.neumann.recordingtools&hl=en_US&gl=US Google Play Store
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No, I wouldn't blame a recording error on anyone but myself. If anything I considered not saying anything but was honest about it and just shared the info to show what we are up against getting good recordings. So many stupid things to go wrong, and I'm not the most fastidious taper around. I would have started the recording but the last one I screwed up was due to batteries running out in the F3 and I'm not 100% confident there. I ended up with 2 bars left on the battery meter after recording just the second set using my Powerex AAs.
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FYI, the F3 looks like it is recording when it isn't. Make sure the red light is on.
Is this behavior common to all Zoom recorders? It is to the F8 (the only Zoom I have much on-hands experience with), which remains in essentially the equivalent of record pause mode whenever it is turned on and not recording or playing, with phantom power on and meters bouncing if mics are attached. Difference may be that on the F8 it's pretty obvious when actually recording as the record button becomes illuminated solid red, in addition to the indication of recording on the display.
I notice this all the time because I rarely disconnect my pre-rigged multichannel microphone array from the recorder. Both occupy the same bag, hooked up at all times. So when playing back files directly from the recorder through the car stereo, head-phones or whatever other playback system I may plug into, as soon as playback ends or is stopped the mics immediately become active.
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FYI, the F3 looks like it is recording when it isn't. Make sure the red light is on.
Is this behavior common to all Zoom recorders? It is to the F8 (the only Zoom I have much on-hands experience with), which remains in essentially the equivalent of record pause mode whenever it is turned on and not recording or playing, with phantom power on and meters bouncing if mics are attached. Difference may be that on the F8 it's pretty obvious when actually recording as the record button becomes illuminated solid red, in addition to the indication of recording on the display.
I notice this all the time because I rarely disconnect my pre-rigged multichannel microphone array from the recorder. Both occupy the same bag, hooked up at all times. So when playing back files directly from the recorder through the car stereo, head-phones or whatever other playback system I may plug into, as soon as playback ends or is stopped the mics immediately become active.
The F6 illuminates the button solid red in the same way. I still can't wrap my head around the physical interface choices on the F3.
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The recorder is as small and stripped down as it can be. I'm attributing any problems to my inexperience with it, and I'm sure I'll be fine. Interesting that the F8 also goes straight to phantom power and meters going. That is certainly not the case with the M10 which I'm used to, which lights up but does nothing until you hit record. Running phantom on powerup is not what I would choose- I'm sure it's draining the battery even when not recording.
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Yes. F8 does has a menu function that saves power by turning off phantom power while playing a file (which I have engaged), but yeah, when simply "idling" it is powered up with phantom sent to whichever channels has it engaged.
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Could someone please post some more Zoom F3 samples? All of the links from earlier in this thread have expired.
Cheers!
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Could someone please post some more Zoom F3 samples? All of the links from earlier in this thread have expired.
Cheers!
Joey DeFrancesco @ Berk's Jazz Fest using Line Audio CM4 (48v) from FOB, DFC ~Wide ORTF mounted to a stationary >:D "object" direct into the F3 @ 32, 48
https://we.tl/t-Y5uoqO0IKo
I'll repost some others but I have to dig for the files on another PC.
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Plenty of dynamic range there - including some handy silences in the first minute - kind of them to play that piece to test your system!
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https://www.sounddevices.com/audio-recording-calculator/
A handy link that's certainly somewhere on TS.
A 32 GB SD card can get roughly 20 hours of 2 tracks of 32 bit audio.
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Hey there, any thoughts on these for the F3? I am tired of burning up AA's and do not want an external battery for this one. TIA https://www.amazon.com/dp/B08QZ5CTC9?psc=1&smid=ADMAJ1NNAHZG9&ref_=chk_typ_imgToDp
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Any reason to choose these over Eneloops? That's what I'm using in my F6.
Fran
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I was looking at the 3400 mah claim on them. How many hours do you get on eneloopes in your F6 BTW, I have one as well. Thanks
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Same cells as Eneloop Pro, but much cheaper:
https://www.ikea.com/us/en/p/ladda-rechargeable-battery-hr06-aa-1-2v-50504692/ (https://www.ikea.com/us/en/p/ladda-rechargeable-battery-hr06-aa-1-2v-50504692/)
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I was looking at the 3400 mah claim on them. How many hours do you get on eneloopes in your F6 BTW, I have one as well. Thanks
You need to be careful on the lithium-ion rechargeable AA cells. The capacity is shown in milli-WATT-hours (mwh), not the usual milli-AMP-hours (mah) shown on a typical nimh cell. If you do the math, those 3400mwh cells are 2267mah at 1.5 volts. These lithium-ion cells have dc-dc converters in them that maintain a constant 1.5 volts until the cell is almost exhausted, then steps the voltage down to trigger low battery warnings in the device being powered.
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I was looking at the 3400 mah claim on them. How many hours do you get on eneloopes in your F6 BTW, I have one as well. Thanks
You need to be careful on the lithium-ion rechargeable AA cells. The capacity is shown in milli-WATT-hours (mwh), not the usual milli-AMP-hours (mah) shown on a typical nimh cell. If you do the math, those 3400mwh cells are 2267mah at 1.5 volts. These lithium-ion cells have dc-dc converters in them that maintain a constant 1.5 volts until the cell is almost exhausted, then steps the voltage down to trigger low battery warnings in the device being powered.
Ahh see, so you get a nice steady voltage, with a little less overall capacity, and then your recording stops and you loose everything? haha. I usually have good luck with amazon's choice stuff... My goal with these is to use 1 set of batteries per set of music, and then recharge, repeat..
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I was looking at the 3400 mah claim on them. How many hours do you get on eneloopes in your F6 BTW, I have one as well. Thanks
You need to be careful on the lithium-ion rechargeable AA cells. The capacity is shown in milli-WATT-hours (mwh), not the usual milli-AMP-hours (mah) shown on a typical nimh cell. If you do the math, those 3400mwh cells are 2267mah at 1.5 volts. These lithium-ion cells have dc-dc converters in them that maintain a constant 1.5 volts until the cell is almost exhausted, then steps the voltage down to trigger low battery warnings in the device being powered.
That's a great tip. Very tricky of them to label them this way to fool the customer into thinking they are getting way higher capacity than they actually are.
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I found an interesting review with comparison against the Eneloops.
And a peek into it.
https://m.youtube.com/watch?v=L62tCyOP06w
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I was looking at the 3400 mah claim on them. How many hours do you get on eneloopes in your F6 BTW, I have one as well. Thanks
Sorry, I haven't done a rundown test that I can recall. I have done shows that were 2 hours without getting nervous.
Fran
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Well, as amazon would have it, I have a 4 pack in my hands already. I charged them, and am gonna throw them in and see how long I can record for tonight or tomorrow before they die. I am curious.
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Well, I tested the batteries with my actives last night. AKG CK1X>Naiant Actives>F3 recording road/ambient noise outside, not sure if SPL's effect drain at all. It was up and running at 4 hours, and was not at 4.5 hours. So for me, one set=1 Set , 2 sets=2sets. A festival if I paid attention, and had a good sized battery to charge them from would be fine with 4 batteries I think as they charge in about 3 hours... Or I could just run off usb-c
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I got an Energizer UE 20068 20AH charger pack, not high enough amperage to power a laptop but a lot of juice. I tested at High Sierra. The first night left me aghast- 2 sets and it was down to 26%. But then for the rest of the festival after I charged it all the way up it only bumped down to 86% at the lowest for the remainder of the sets and topping off a couple of times. I'd charge my phone off of it too and that didn't make a dent. So I think that first night was a faulty first charge up and things will be fine after this. It takes quite a few hours to charge up. The USB C connector did not come out and you can tell if it does because then the battery charge icon comes on. There is no icon to indicate external power is good- if your screen is up and there's no icon that is your signal that external power is working. I got it because I didn't have time to wait and I oppose Bezos's take over of the world.
https://www.bestbuy.com/site/energizer-max-20000mah-high-speed-universal-portable-charger-power-bank-with-lcd-display-for-apple-android-google-usb-devices-black/6498909.p?skuId=6498909 (https://www.bestbuy.com/site/energizer-max-20000mah-high-speed-universal-portable-charger-power-bank-with-lcd-display-for-apple-android-google-usb-devices-black/6498909.p?skuId=6498909)
As far as batteries with internal charging circuitry, I'd not trust that. I charge my AAs and AAAs with a Powerex charger that you can program the charge/discharge rate and cycle twice to get them perfectly topped off. When you do that you know what's in it because the charger put it there and kept track of it. My program is: Cycle>700 charge>500 discharge> 2 cycles. I've tried single cycle and don't get the same amp hours loaded up. The numbers are remarkably consistent but it tends to call out cheap AAs as junk. That hasn't happened with better batteries.
https://www.bhphotovideo.com/c/product/1572727-REG/powerex_mh_c9000pro_c9000pro_professional_charger_analyzer.html (https://www.bhphotovideo.com/c/product/1572727-REG/powerex_mh_c9000pro_c9000pro_professional_charger_analyzer.html)
I had zero recording errors for the festival and got it looks like about a dozen sets recorded. There's a bunch up on Archive, and the rest of the team did a good job as well.
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I tested a set of eneloops and only got about 2.25 hours using a pair of Schoeps mk21's with 48V phantom used bluetooth to set the recording then it lost connection. Second try... only 24 minutes with the beyer ck930s and the cv900 preamps + the bluetooth on monitoring.
I have an Ikea Solbana 10000 mAh power bank that is just barely too wide to fit into the back of this unit that I'll probably run with it. *Update on the Solbana, it ran the power hungry ck930s with bluetooth on for just under 17 hours.
Has any progress been made to remove the metal frame on the front?
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I have not made another attempt. I've been a bit busy and I haven't needed it for a stealth unit yet. Encouraging others to try but understand if you don't want to. I safely got mine to where the two halves were separating but catching on the battery spring, and at that point there would probably be some connectors and wires to fold/stuff. Hopefully then some screws to unfasten the frame would present themselves.
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I get 4 Hours with eneloop PRO, with two SE8 mics :)
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Are there any decent mini cable free mics that plug directly into the XLR ports?
I recently got my F3 and an Audio-Technica BP4025 stereo mic and LOVING it, but it's a little slow to setup, especially for birding. An all in one setup would be great for quick recordings before the bird flies away.
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This little recorder is just crying out for a dedicated plug-in stereo mic. There has been discussion on several forums about this idea. There is the CEntrance PM1 mic set (less than 200 bucks) designed for their own recorder. They will fit the F3, but due to the orientation of the XLR plug, you can’t get an X/Y orientation, just an A-B. Also, the PM1 has a self-noise of 20db(A). Comparing that to the BP4025s 14db(A), you will hear more noise (hiss) in the background. Whether this is acceptable will depend on how quiet the source audio you are trying to capture is. 20db(A) is too noisy for nature recording, IMHO. Maybe some company will make a stereo mic head with low noise cardioid capsules like the 4025, plus an attached pair of XLR connectors that fit the spacing and orientation of the F3, creating a compact unit. I’m sure there will be some DYI mic designs showing up, given some time.
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At last someone makes a recorder without built in mics - and then...
I'm slightly surprised that it doesn't accept even Zoom's own clip on mics like the F1 does. Maybe they could make a simple adapter which converts the XLR connectors into their own proprietary connector. Or maybe a 3rd party could do that. Then you could connect their MS mic to the front. No, wait a minute, it doesn't support MS mics... :banging head:
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Yeah, I know. This is a tapers forum and stealth is a priority. Apologies for the tangent into the “outside” world! ;)
That zoom proprietary connector is intriguing. The modules clearly have active electronics in them, as well as the mic capsules, so there has to be contacts for supplying power. If an adapter were to be made, the only source of power would be the phantom supplies through the XLRs, and I’m doubtful there is enough there to run the circuits. It would help if there was publicly available info about the modules internals.
It really isn’t true that the F3 doesn’t support mid-side mics, it just doesn’t support monitoring in decoded m-s. I use my F3 with my mid-side mic setup all the time. I just can’t listen to the decoded audio until it gets loaded into my computer later on. I suspect there will be a firmware update along the way that will add mid-side functions (at least monitoring, if not decoded recording). I’m sure Zoom is well aware of it by now.
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This little recorder is just crying out for a dedicated plug-in stereo mic. There has been discussion on several forums about this idea. There is the CEntrance PM1 mic set (less than 200 bucks) designed for their own recorder. They will fit the F3, but due to the orientation of the XLR plug, you can’t get an X/Y orientation, just an A-B. Also, the PM1 has a self-noise of 20db(A). Comparing that to the BP4025s 14db(A), you will hear more noise (hiss) in the background. Whether this is acceptable will depend on how quiet the source audio you are trying to capture is. 20db(A) is too noisy for nature recording, IMHO. Maybe some company will make a stereo mic head with low noise cardioid capsules like the 4025, plus an attached pair of XLR connectors that fit the spacing and orientation of the F3, creating a compact unit. I’m sure there will be some DYI mic designs showing up, given some time.
I wouldn't be so sure that mics with 20 dB self noise are unsuitable. I don't do nature recording, but I do classical recording all the time with my 4061s which are rated 26 dB self noise, and the mics never draw attention to themselves. Many others who use the 4060/4061 as their main pair can attest to this. The noisefloor of your venue, even if it is a quiet concert hall or secluded location in the woods, is likely going to have its own noisefloor.
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Yep. Depends on the situation. I use DPA 4060 with a self-noise a few dB lower than 4061 for classical and nature recording and have never had a mic self-noise problem with them for any live classical performance recording. I have noticed self-noise at times in some nature recording situations. Difficult to compare though, as some nature recordings tend to be used at higher than normal playback levels making the recording's noise-floor apparent regardless of its source, due to a limited overall dynamic range and everything happening down at the quiet end of that range. Sure, other nature recordings exhibit extreme dynamics (close thunder storm recordings for example) and thus cannot be amplified as much (without limiting or compression), so the the noise floor remains below perception. Of course, classical recordings vary widely in dynamics too (delicate small ensemble chamber music vs big bombastic symphonic works) but in general tend not to be played back significantly louder than the original performance.
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I'm slightly surprised that it doesn't accept even Zoom's own clip on mics like the F1 does.
Not having used the F3 myself, I just assumed it would had the capability to accept the same directly-attachable Zoom mics that are compatible with all other F-series recorders.
That said, I've no idea of the quality of the directly-attachable Zoom microphones.
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This little recorder is just crying out for a dedicated plug-in stereo mic. There has been discussion on several forums about this idea. There is the CEntrance PM1 mic set (less than 200 bucks) designed for their own recorder. They will fit the F3, but due to the orientation of the XLR plug, you can’t get an X/Y orientation, just an A-B. Also, the PM1 has a self-noise of 20db(A). Comparing that to the BP4025s 14db(A), you will hear more noise (hiss) in the background. Whether this is acceptable will depend on how quiet the source audio you are trying to capture is. 20db(A) is too noisy for nature recording, IMHO. Maybe some company will make a stereo mic head with low noise cardioid capsules like the 4025, plus an attached pair of XLR connectors that fit the spacing and orientation of the F3, creating a compact unit. I’m sure there will be some DYI mic designs showing up, given some time.
I wouldn't be so sure that mics with 20 dB self noise are unsuitable. I don't do nature recording, but I do classical recording all the time with my 4061s which are rated 26 dB self noise, and the mics never draw attention to themselves. Many others who use the 4060/4061 as their main pair can attest to this. The noisefloor of your venue, even if it is a quiet concert hall or secluded location in the woods, is likely going to have its own noisefloor.
I agree that the 4060/4061s are great sounding mics. I believe audibility of self-noise depends on the amount of masking the intended sound source provides. I admit to being spoiled; my main soundscape recording rig uses Sennheiser MKH8020s with a self-noise of 10db(A). I have a pair of Earthworks QTCs (21db(A)) that I’ve tried for nature recording. They sound great, but the noise is intrusive. I hate using noise reduction software, so I stick with the Sennheisers (which work great with the F3, BTW)
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This little recorder is just crying out for a dedicated plug-in stereo mic. There has been discussion on several forums about this idea. There is the CEntrance PM1 mic set (less than 200 bucks) designed for their own recorder. They will fit the F3, but due to the orientation of the XLR plug, you can’t get an X/Y orientation, just an A-B. Also, the PM1 has a self-noise of 20db(A). Comparing that to the BP4025s 14db(A), you will hear more noise (hiss) in the background. Whether this is acceptable will depend on how quiet the source audio you are trying to capture is. 20db(A) is too noisy for nature recording, IMHO. Maybe some company will make a stereo mic head with low noise cardioid capsules like the 4025, plus an attached pair of XLR connectors that fit the spacing and orientation of the F3, creating a compact unit. I’m sure there will be some DYI mic designs showing up, given some time.
I wouldn't be so sure that mics with 20 dB self noise are unsuitable. I don't do nature recording, but I do classical recording all the time with my 4061s which are rated 26 dB self noise, and the mics never draw attention to themselves. Many others who use the 4060/4061 as their main pair can attest to this. The noisefloor of your venue, even if it is a quiet concert hall or secluded location in the woods, is likely going to have its own noisefloor.
I agree that the 4060/4061s are great sounding mics. I believe audibility of self-noise depends on the amount of masking the intended sound source provides. I admit to being spoiled; my main soundscape recording rig uses Sennheiser MKH8020s with a self-noise of 10db(A). I have a pair of Earthworks QTCs (21db(A)) that I’ve tried for nature recording. They sound great, but the noise is intrusive. I hate using noise reduction software, so I stick with the Sennheisers (which work great with the F3, BTW)
Thanks everyone for the ideas! I'm definitely not a fan of noise, so 20db(A) would be out of the question. While perusing Google Images for mics I ran across the Wildtronics Micro Mic XLR Microphone and ordered one for $91.60 shipped. Small, low noise 14db(A) and it looks like it'll plug directly in for quick mono recordings. I can switch to my BP4025 if time permits. I'll let you know how it works out. https://www.wildtronics.com/micro-xlr.html (https://www.wildtronics.com/micro-xlr.html)
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Not having used the F3 myself, I just assumed it would had the capability to accept the same directly-attachable Zoom mics that are compatible with all other F-series recorders.
That said, I've no idea of the quality of the directly-attachable Zoom microphones.
That connector is absent on the F6 as well. They could have fit it, but instead gave it an NP-F battery connector (which was a great decision, IMHO).
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Agreed. I didn't know the F6 lacks it.. and am surprised the NP-F battery connector change didn't carry over to the F8nPro. I can't see many users using the Zoom mic modules with an F8, even with the cable extender, and expect most would prefer an NP-F battery connector. I could see it being useful if the channels though that connector could be recorded to the stereo mix channel instead of an internal mix, effectively making the recorder capable of recording 10 inputs (like the Tascam's DR-680, who's digital input can be routed to the stereo channel, effectively making that recorder capable of recording 8 input channels)
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Indeed I'm aware that the missing element is MS monitoring, as you can actually record MS on anything with two channels, but for me in the context of classical music recording, placement of MS mics is critical (well, as with any other mic technique) and if I can't check that I've got it correct using dematrixed rehearsal playback, then it's a problem. It's really surprising to me that this simple thing was omitted on release as it's a common technique for film/tv which this device seems to be marketed for.
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Indeed I'm aware that the missing element is MS monitoring, as you can actually record MS on anything with two channels, but for me in the context of classical music recording, placement of MS mics is critical (well, as with any other mic technique) and if I can't check that I've got it correct using dematrixed rehearsal playback, then it's a problem. It's really surprising to me that this simple thing was omitted on release as it's a common technique for film/tv which this device seems to be marketed for.
You have mentioned this before - have you contacted Zoom tech support asking if it can be included in a firmware update?
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I've also mentioned that here with regard to the F8, and have emailed Zoom support a couple times about it's M/S implementation deficiencies, but never heard back from them. On the F8 it works fine for monitoring while recording, but is far from ideal for playback. If one has previously recorded raw Mid and Side channels, M/S playback matrixing to L/R is routed post-fader, with no ability to adjust playback level or ratio. Wish they would fix that.
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I wrote to Zoom customer service in the US about adding a simple M-S decoding mode to the F3 headphone monitor signal path. I received a response within a few hours stating they would forward the request to the software engineers in Japan for consideration in an upcoming firmware revision. I believe they are responsive, but only time will tell if the people that decide these things hear and act on that request.
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Last week, I finally got the Zoom F3 recorder.
Now my AT822 mic picks up powerful, solid and detailed sound. :-)
However, I connected the Zoom F3's Line out and Sony a7R IV camera's Mic in with a -25db attenuator cable, but only buzz sound came out.
Do I need a stronger attenuator cable like -35db?
Your input would be appreciated.
Thanks.
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The Zoom F3 has a built-in line output attenuator that can go as low as -48db. No attenuator cable should be necessary. Reading the (Version 2.0) manual from page 41 describes the setup process along with a way to calibrate the gain structure.
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Yes, I know. However, even when it is reduced to -48db, only a strange sound comes out of the Sony headphone jack.
I tried connecting the line out of the Zoom F3 to the line in of the R-05 recorder and it worked. ???
Thanks.
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I wonder if the mic input on the camera has plug-in-power enabled? The voltage present on the jack terminals could be upsetting the output of the F3. I could not quickly find a manual for your camera, but there should be a menu setting for plug-in-power (P-I-P), in the external audio input section. I can’t think why else there would be an issue, assuming the camera mic input is a standard stereo 3.5mm input jack.
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I found the cause of the problem.
- https://www.amazon.com/Woisco-Charging-15000mAh-Portable-External/dp/B09K42VFN5/
I connected USB-C and USB-A from the RAVPower 15,000 mAh USB-C Portable Power Bank to power the Zoom recorder and Sony camera.
The strange noise comes when both devices are powered from the same power bank.
However, when only one device is connected, there is no noise. :P
Thanks for all the replies.
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I wrote to Zoom customer service in the US about adding a simple M-S decoding mode to the F3 headphone monitor signal path. I received a response within a few hours stating they would forward the request to the software engineers in Japan for consideration in an upcoming firmware revision. I believe they are responsive, but only time will tell if the people that decide these things hear and act on that request.
There was a bug in Zoom's Ambisoncs Player for the H3-VR with the rear channels missing on playing back B-format as 5.1. Having had the experience that their German importers are totally hopeless, I contacted Zoom USA. They asked for example files and screenshots, told me they'd be forwarding everything to Japan, and a few weeks later informed me that there was a new version of the software and would I please check. I checked and everything was fine. Then I looked in the version history and found that they had issued a new release for one single problem... my bug report. Now, if that isn't responsive I don't know what would be.
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Newsshooter just published a comprehensive review of the F3 that might be useful for presumptive buyers.
https://www.newsshooter.com/2022/08/02/zoom-f3-review/
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I wrote to Zoom customer service in the US about adding a simple M-S decoding mode to the F3 headphone monitor signal path. I received a response within a few hours stating they would forward the request to the software engineers in Japan for consideration in an upcoming firmware revision. I believe they are responsive, but only time will tell if the people that decide these things hear and act on that request.
There was a bug in Zoom's Ambisoncs Player for the H3-VR with the rear channels missing on playing back B-format as 5.1. Having had the experience that their German importers are totally hopeless, I contacted Zoom USA. They asked for example files and screenshots, told me they'd be forwarding everything to Japan, and a few weeks later informed me that there was a new version of the software and would I please check. I checked and everything was fine. Then I looked in the version history and found that they had issued a new release for one single problem... my bug report. Now, if that isn't responsive I don't know what would be.
Very good to hear. This gives me reason to hope the mid-side feature will appear in a future firmware update.
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Newsshooter just published a comprehensive review of the F3 that might be useful for presumptive buyers.
https://www.newsshooter.com/2022/08/02/zoom-f3-review/
Very informative. I need to put my hands on it. I do not want XLRs since it is just something else to get in but most of the other things are appealing. The UI would take an adjustment as well.
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How is the headphone out? Strong enough to drive high impedance cans?
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Mine arrived today from B&H. Excited to try it out. Thinking of getting a really short pair of angled interconnects and just leave them in, to minimize the wear on deck. What other tricks have you picked up. Anyone got a shot of the gaffer fix for the hold?
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Got a Zoom F3 this week. I will do some battery-tests and post the results here. In the meantime,
- If you loose the Bluetooth connection while recording there is no way to get it back is it? (unless you stop the recording and reconnect.) Now in the testsituation i was recording and moving away further away from the unit. In the field this might not happen very soon.
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Seems like I am losing Bluetooth connection fast. Are more people having this problem?
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Seems like I am losing Bluetooth connection fast. Are more people having this problem?
short answer, yes.
I find that if I turn my phone screen off for too long, it'll drop, if I turn weird, it'll drop, if I sneeze, probably drop.
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Seems like I am losing Bluetooth connection fast. Are more people having this problem?
short answer, yes.
I find that if I turn my phone screen off for too long, it'll drop, if I turn weird, it'll drop, if I sneeze, probably drop.
Probably Corona... :shrug:
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that's unfortunate. the F6 bluetooth is rock solid
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Recordingstests
Microphones - Recording Time - Batterybrand - Capacity
Nevaton MC59/S - 2h 46m - eneloop - 1900mAh
Milab VM44-Link - 2h 04m - eneloop - 1900mAh
Nevaton MC59/S - 4h 04m - ansmann - 2850mAh
Milab VM44-Link - 4h 13m - ansmann - 2850mAh
Mic consumption current from Milab is 4mA and Nevaton is 5mA.
Factorysettings, except: Mic 48+ / Bluetooth OFF / LCD backlight 1 min on / Contrast 1 (of 10)
The effect on Bluetooth on batterylife could not be tested as BT loses connection very fast. That is the real bad thing from the Zoom F3 if you need it. Have not found any other issues. Just a nice small 48v brick that records too.
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Mine arrived today from B&H. Excited to try it out. Thinking of getting a really short pair of angled interconnects and just leave them in, to minimize the wear on deck. What other tricks have you picked up. Anyone got a shot of the gaffer fix for the hold?
I do not have any issues with the hold button. It has to be pushed up twice to turn off, once will not turn it off. I have recorded a lot of shows with this deck and have not had any issues. It has changed much of how I record low pro, as I find it foolproof.
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Mine arrived today from B&H. Excited to try it out. Thinking of getting a really short pair of angled interconnects and just leave them in, to minimize the wear on deck. What other tricks have you picked up. Anyone got a shot of the gaffer fix for the hold?
I do not have any issues with the hold button. It has to be pushed up twice to turn off, once will not turn it off. I have recorded a lot of shows with this deck and have not had any issues. It has changed much of how I record low pro, as I find it foolproof.
What is your opinion on the Bluetooth failing, does this happen to you too?
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Mine arrived today from B&H. Excited to try it out. Thinking of getting a really short pair of angled interconnects and just leave them in, to minimize the wear on deck. What other tricks have you picked up. Anyone got a shot of the gaffer fix for the hold?
I do not have any issues with the hold button. It has to be pushed up twice to turn off, once will not turn it off. I have recorded a lot of shows with this deck and have not had any issues. It has changed much of how I record low pro, as I find it foolproof.
What is your opinion on the Bluetooth failing, does this happen to you too?
I bought the Bluetooth add on, but have never used it. I just hit record and if it is low pro and I hit it early, and I chop off the excess in post. No levels to worry about so why should I worry about starting the deck on time or having it fail because the Bluetooth has disconnected? When I tested it the Bluetooth seemed robust, but I just like the fact that this is a true set and forget deck, so I have not used or messed with the Bluetooth add on during a show.
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Recordingstests
Microphones - Recording Time - Batterybrand - Capacity
Nevaton MC59/S - 2h 46m - eneloop - 1900mAh
Milab VM44-Link - 2h 04m - eneloop - 1900mAh
Nevaton MC59/S - 4h 04m - ansmann - 2850mAh
Milab VM44-Link - 4h 13m - ansmann - 2850mAh
impressive! whats the catch, that they lose chaarge quickly in storage? the amazon link says 2% per day loss which is several times higher than a normal non-lsd nimh
do they charge the same as any other nimh?
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Mine arrived today from B&H. Excited to try it out. Thinking of getting a really short pair of angled interconnects and just leave them in, to minimize the wear on deck. What other tricks have you picked up. Anyone got a shot of the gaffer fix for the hold?
I do not have any issues with the hold button. It has to be pushed up twice to turn off, once will not turn it off. I have recorded a lot of shows with this deck and have not had any issues. It has changed much of how I record low pro, as I find it foolproof.
What is your opinion on the Bluetooth failing, does this happen to you too?
I bought the Bluetooth add on, but have never used it. I just hit record and if it is low pro and I hit it early, and I chop off the excess in post. No levels to worry about so why should I worry about starting the deck on time or having it fail because the Bluetooth has disconnected? When I tested it the Bluetooth seemed robust, but I just like the fact that this is a true set and forget deck, so I have not used or messed with the Bluetooth add on during a show.
Good approach, probably this will how it works in practise. To me, now, feels nice to have a bit of control during a recording, but then again....with this unit, there is nothing to control :)
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Recordingstests
Microphones - Recording Time - Batterybrand - Capacity
Nevaton MC59/S - 2h 46m - eneloop - 1900mAh
Milab VM44-Link - 2h 04m - eneloop - 1900mAh
Nevaton MC59/S - 4h 04m - ansmann - 2850mAh
Milab VM44-Link - 4h 13m - ansmann - 2850mAh
impressive! whats the catch, that they lose chaarge quickly in storage? the amazon link says 2% per day loss which is several times higher than a normal non-lsd nimh
do they charge the same as any other nimh?
I use a Powerex charger to charge my nihms. That thing has a lot of feature, like uncharge (I hardly know how to deal with), but i just plug them in and let it charge. I can not be bother with info on losing charge time in storage and all. Using batteries out of storage without checking is just not something to do for me. I just charge them on the day of the concert so I know they are full. If you want more juice, I recommend a powerbank from Anker.
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Good approach, probably this will how it works in practise. To me, now, feels nice to have a bit of control during a recording, but then again....with this unit, there is nothing to control :)
A good recording is just as much about mic placement, location, mic selection and so on. That is also much of the fun of it. I have no guilt at all after 50 years of recording to not worry that my level is too hot or too low. It's all about the end product after all.
;D
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The first concert recording with my friend's F3 did not go smoothly. As epic write failures go, this is one of the worst I've heard.
https://taperssection.com/index.php?topic=200929.0 (https://taperssection.com/index.php?topic=200929.0)
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^ Any idea about the provenance of the card? It is my understanding that there are tons of fake cards out there. I am skittish about this to the extent that I only buy cards from big retailers, cost be damned...
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^ Any idea about the provenance of the card? It is my understanding that there are tons of fake cards out there. I am skittish about this to the extent that I only buy cards from big retailers, cost be damned...
Yes, there are a lot of counterfeits out there. He sent me a screenshot from Amazon, but I can't say for sure if he used Amazon's authorized SanDisk store or a third-party seller.
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^ Hopefully, it is just a bad card. It sucks to pull a recording onto the computer and see that it is messed up so badly...
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^ Hopefully, it is just a bad card. It sucks to pull a recording onto the computer and see that it is messed up so badly...
We're performing the same concert again in 2 weeks, but in a venue with much better acoustics. I already planned to run a 4-mic array there of my CM3s and 4061s, and I'll just add his Blumlein 414s to the last two channels of my F6. Maybe do line out of just his 414s into the F3 as a backup.
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^ That would be a great opportunity to try a new card in the F3! It seems unlikely that the recorder is at fault, since there haven't been any reports (that I have seen) about issues and I think there are quite a few out there. Lucky to have a second chance, too.
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^ That would be a great opportunity to try a new card in the F3! It seems unlikely that the recorder is at fault, since there haven't been any reports (that I have seen) about issues and I think there are quite a few out there. Lucky to have a second chance, too.
Yep. I was actually going to tell him to get a different card from the list regardless of what my testing showed. Maybe one of the Kingston Canvas Select cards - I have had flawless performance with those in my F6.
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Don’t forget that the F3 has an internal SD card test function that could help root out incompatible cards before committing important recordings to them. I’ve had zero card read/write issues with my F3 and a Zoom recommended card.
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Don’t forget that the F3 has an internal SD card test function that could help root out incompatible cards before committing important recordings to them. I’ve had zero card read/write issues with my F3 and a Zoom recommended card.
Just did that this morning, and it passed. Also reformatted. Right now I'm letting a recording run for a few hours to see if I can replicate the issue.
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prior to the error was the card formatted in the F3 or on another device/computer?
just thinking out of the box in regard to all possible culprits
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prior to the error was the card formatted in the F3 or on another device/computer?
just thinking out of the box in regard to all possible culprits
It's not my recorder or card, so I don't know what he did with it.
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Bummer, for what it is worth,
https://www.amazon.com/Transcend-TS32GUSD300S-UHS-I-MicroSD-Memory/dp/B07CD9QYNM
This I have used in the field and sofar no problem. It is on the tested list from Zoom.
I remember from earlier threads from different recorder that it is wise to format in the recorder prior to recording. Something I always do.
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Bummer, for what it is worth,
https://www.amazon.com/Transcend-TS32GUSD300S-UHS-I-MicroSD-Memory/dp/B07CD9QYNM
This I have used in the field and sofar no problem. It is on the tested list from Zoom.
I remember from earlier threads from different recorder that it is wise to format in the recorder prior to recording. Something I always do.
Thanks for the tip on that card. Crazy how cheap they have gotten.
Yes, I know to format in the recorder, but my friend that actually owns the F3 and is brand new to all this probably didn't know that.
In any case, I did a 2-hour test recording last night with his card freshly formatted and fully tested, and it had the same buffer underrun recording skips as the concert recording. I had him order the 128 GB Kingston card off the list, and we'll do a test with that one. Hopefully it's just the card and his F3 isn't borked.
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In any case, I did a 2-hour test recording last night with his card freshly formatted and fully tested, and it had the same buffer underrun recording skips as the concert recording.
Same recorder for the test or a different one?
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In any case, I did a 2-hour test recording last night with his card freshly formatted and fully tested, and it had the same buffer underrun recording skips as the concert recording.
Same recorder for the test or a different one?
Same recorder; same card. I send the F3 back to him last night and he's going to do a long recording when the new card gets there.
I was going to run all 6 of our mics into my F6 for the concert in two weeks and use his F3 as a backup, but I just discovered there is no TRS input. I have all the connectors to solder up an adapter cable, but not sure it's worth the trouble.
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Bummer, for what it is worth,
https://www.amazon.com/Transcend-TS32GUSD300S-UHS-I-MicroSD-Memory/dp/B07CD9QYNM
This I have used in the field and sofar no problem. It is on the tested list from Zoom.
I remember from earlier threads from different recorder that it is wise to format in the recorder prior to recording. Something I always do.
Here's another card I've had good luck with so far. It's also UHS-II which makes for very fast transfers in a UHS-II card reader.
ProGrade Digital 128GB UHS-II microSDXC Memory Card with SD Adapter
https://www.bhphotovideo.com/c/product/1560273-REG/prograde_digital_pgmsd128gbpbh_128gb_microsdxc_uhs_ii_memory.html (https://www.bhphotovideo.com/c/product/1560273-REG/prograde_digital_pgmsd128gbpbh_128gb_microsdxc_uhs_ii_memory.html)
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Update on my friend's F3:
After I had him get a different card on the tested list, he ran into the same problems. After contacting Zoom support, they had him send it in. His unit was in fact defective, and they send him a replacement that has been working perfectly.
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Thinking about getting one of these for Jazzfest taping this year. I like the idea of not having to do any level setting just get in place and hit record.
Any thoughts on the practicality of it for running in a pocket? A couple of the shows I plan to tape are seated in the Jazz Tent with mics in a hat. The rest are out in the open on a stand out of doors.
Probably be using it with Line Audio CM3s or maybe MBHO KA500 if I have that setup built up by then.
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You'll need a good deep pocket to really fit it comfortably. I've often times just left it in a coat jacket since it's profile stands so high with the belt loops on the back.
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Thinking of picking up one for shits and gigs, but if it can’t get 3 hours on Powerex 2650 then not bueno. Anybody do a battery test?
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Thinking of picking up one for shits and gigs, but if it can’t get 3 hours on Powerex 2650 then not bueno. Anybody do a battery test?
You will get 3-4 hours depending in your mics. I just had one die this weekend that was running phantom at the 3 hours, 6 minutes at 32/48 with display ON. deleepow 3400mWh YMMV I need to repeat test with display OFF.
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I get over 8 hours on a set of Energizer lithiums.
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I get over 8 hours on a set of Energizer lithiums.
That is great to know. Display on or off out of curiosity and Phantom? I tested this before with alkalines and got just under 4 hours. The lithiums you speak of are throwaways? Just curious. I run an outboard pack when there is any question of set length.
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you can use usb-c power to overlap while you swap rechargeables too
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I get over 8 hours on a set of Energizer lithiums.
That is great to know. Display on or off out of curiosity and Phantom? I tested this before with alkalines and got just under 4 hours. The lithiums you speak of are throwaways? Just curious. I run an outboard pack when there is any question of set length.
Yes, the throwaways. Display on, phantom on running Schoeps capsules with a pfa.
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I plan to make a detailed thread with pictures shortly but I did go to the next level in taking the thing apart. I think maybe we can ditch this giant hunk of metal, possibly.
Anyway, I pulled the back half of the plastic clamshell off past the battery spring, which wasn't actually too difficult.
Then I still didn't see the mount screws for the metal part so I went the next step and pulled the mic input subassembly. This slid out pretty easily, but at this point there's one of those plastic trace cables which are oh so delicate. I'd broken one on a point and shoot in the past, and I have to say I hate them. So this tucks in oddly on the side and pulling the mic sub out is likely to damage it. Be super careful at this point and if you try to do what I'm doing take some pictures. I did see two of the mount screws for the metal frame at this point but the other two presumably are under the main circuit board. This is where I gave up on this try and will try again possibly, though I'm close to breaking it if I haven't already. Maybe the thing to do would be to pull out the plastic trace cable from the mic sub before sliding it out.
Where do people host pictures for this board? Just your Google Drive? Photobucket is kind of defunct.
(https://drive.google.com/file/d/1Tjavu0dOsWOktRGOk-IXisP9VwWvlKKl/view?usp=share_link)
I think you can get 3 hours out of it with Powerex but not with a comfortable margin of error. I recorded Midnight North and then Phil without powering down during set breaks, missed most of Phil's second set. You can bring a phone charger unit for the hot swap or additional time. I mean to test what happens when the external runs out. My 20AH Energizer pack goes for days.
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You can post pictures directly. Under the box for typing a reply, it says "Attachments and other options". Click that and you get a "Choose File" button.
"Allowed file types: bmp, gif, jpg, jpeg, mp3, ogg, pdf, torrent, txt, shn, wav, zip
Restrictions: 10 per post, maximum total size 7500KB, maximum individual size 750KB"
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Clamshell coming apart after removing the 4 tiny screws from the back and two of the XLR mount screws. At this point you need a screwdriver under one of the battery springs.
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Pulling out the mic input sub I didn't pay attention enough and this was the result. Do this slowly and keep an eye on this attachment cable. I assumed it would be some wire cables to the main. You can see the two screws that I think hold the metal frame on. Presumably there are 2 more under the main board. It would have been nice to have these removeable from the front of the unit and then we could have just slid it off.
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It took a bit of prodding and stuffing but I got this back where it belonged and put it back together.
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Another view of the difficult connection, showing where maybe to pull out the connector next time.
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I just had the idea that if I'm successful in getting the cover removed, to scan/measure it and print a modified version in ABS to keep the protection.
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Wow. A lot of work taking this apart.
I've been using one of these for a while. I really like it. Simple and reliable.
The noise floor is low and there appear to be no (noticeable) artifacts in the spectrum.
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Thanks for sharing this teardown - very interesting! Just around a year of ownership with this thing and it's proving to be my main >:D go-to device. As I noted somewhere in this thread, I've got this thing into a Hancock show at the Kennedy Center with minimal trouble, so it's proven itself beyond useful, particularly for >:D.
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Just bought an F3. Recommendations regarding card capacity? Does it really handle 1 TB cards well? Thanks in advance for any guidance.
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Just bought an F3. Recommendations regarding card capacity? Does it really handle 1 TB cards well? Thanks in advance for any guidance.
I use a Sandisk 256GB card with no problems at all. I can't imagine a situation where I would need a 1TB card.
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OK the latest episode of everyone's favorite fun game, burn $350! Anyway, hopefully that's not the outcome here but I think I've hit pay dirt. Now the question of what next.
I paid more attention to the flat cable this time. I tried to pull the connector out of the mic input board but gave up on that. Later, the traces pulled out of the connector and I pushed them back in, hopefully all is still good there. First photo.
Then, unscrew the usb connector. Second photo.
Slide switch in the orientation it needs to be for re-assembly. 3
Bluetooth plug out. 4
Screws that need to come out. The one in the middle does not, however there is one hidden to the left of the one in the lower left corner, that does need to be taken out. Also the two on the far side that are in the open and more obvious. 5
Now grab the black plastic case by the edges and lift out. Use care to keep the flat traces to the mic input card intact. 6
The buttons stick with the case, but are not fastened. They can be laid into the plastic case, need some way to hold these. 7
Behold the giant chunk of metal that we don't need. 8.
It can now be put back together with or without the frame/plate. You can see 9,10, what it will look like. I am leaning towards cutting off the heavy legs of the frame and re-assembling. Quick napkin calc shows this will reduce the weight from 3oz to 1oz of (what feels like) steel. If its aluminum then it would be 1oz down to 1/3 oz. But I'm thinking closer to a QP than a lid? So probably steel.
Next steps? Cut off the uprights to preserve the structural and electrical purposes of the case? 3D print in ABS or some plastic and use a foil lining?
I used the post office scale to weigh the frame. 5.3oz! OVER a quarter pound of steel protecting this bit of goodness.
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Keep going and think of what you'll charge us for the mod :)
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Success! I went and got me a Dremel, finally. I think 57 is old enough to finally be the guy with the Dremel. Note the nickel plated screws go through to the metal, the zinc ones are for the internal stuff. The plastic trace is still the weak point of doing this and probably more attention could be paid to unplugging this before doing anything else to protect it. When I was done I realized I could still slide the legs back on, attached by the belt bars. I might be able to work out how to make this click into place for non-stealth situations. It is still a stout unit, but not the tank that it was.
Here's the final breakdown:
Complete cover and belt bars 5.3 oz
Legs and belt bars 4.3 oz
Recorder with modified face plate (minus batteries) 4.6 oz
By subtraction, the remaining face plate 1.0 oz
By addition, complete recorder 8.9 oz
So the legs constituted a full quarter pound of steel. Almost half of the weight of it. I had to break it down one more time as I missed a screw. I'm getting pretty quick with it, took like 15 minutes now that I know what I'm doing.
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Nice work. Congratulations, you now take possession of the Guy With A Dremel modification award trophy originally held by M0ke.
And have me scheming about the possibilities of similarly stealth-miniaturize modding an F6.
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Eman
With the photos in your last post, does that mean one might be able to just cut the legs away and leave the top holding the screen in place?
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Eman
With the photos in your last post, does that mean one might be able to just cut the legs away and leave the top holding the screen in place?
Yes, I have mounted the remaining plate back which holds the buttons and LED lens in place with 4 mounting screws. Look at the photo of the back of the face plate to see the mount holes I cut around. I don't think you could do it without removing the frame from the recorder if that's what you mean. You would almost definitely eat into the plastic during cutting. Perhaps you are a Jedi master with the Dremel. Use the force. I measured all the features on the back before putting it back together, can give you the distance from the window to each hole.
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Ready for action!
I'm travelling for work in a minute for the week, so won't be able to track out but here's the raw mp3
https://drive.google.com/drive/folders/1NwVzHvqzGhJW1Yqv73vikq14pVcEpb54?usp=share_link
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Nice, can you share a concert somewhere. Would love to hear this combo!
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did a battery runtime test, with some fully charged nimh left to rest overnight
mics: schoeps cmc1 (2mA draw @ 48V)
wasnt paying super close attention, but it went from 2 bars to 1 bar on the battery meter somewhere between 3:40 and 4:05. zero bars/Low battery indication at 4:50, shutoff at 5:06. so like an hour on last bar and 15 minutes after low batt warning
seems more than adequate to set up and start recording well before the show, and roll straight through setbreak. of course not all P48 mics are that efficient
edit: subsequent tests i couldnt get much over 4 hours, in fact with different sets of ~2400-2600 ma batteries, all were between 3:50 and 4:10. I can only conclude i boned reading the clock and thought i gained an hour since i cant reproduce. 4 hours is tight so in an open situation i will be using a small usb battery and using the internal batteries as backup
when the battery indicator changed from 3>2, 2>1, 1>0 bars also had a huge range
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did a battery runtime test, with some fully charged nimh left to rest overnight
mics: schoeps cmc1 (2mA draw @ 48V)
wasnt paying super close attention, but it went from 2 bars to 1 bar on the battery meter somewhere between 3:40 and 4:05. zero bars/Low battery indication at 4:50, shutoff at 5:06. so like an hour on last bar and 15 minutes after low batt warning
seems more than adequate to set up and start recording well before the show, and roll straight through setbreak. of course not all P48 mics are that efficient
nice! which batteries and capacity did you use?
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$275 refurb on Amazon with 90 day money-back guarantee.
Too many recorders, the socks drawer is overflowing.
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$232 as a prime day special on amazon today.
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$232 as a prime day special on amazon today.
Unbelievable price, set it to line in and get that soundboard, it can handle it!
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$232 as a prime day special on amazon today.
Thanks! Just ordered one.
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I grabbed one for $99 after using my Amazon points. Figure it's worth trying with my NBox Platinum for non metal detector shows since I will be using XLR cables and my usual way for bringing gear in may be problematic.
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What sample rate is everyone using? Going to hook it up the the house board tonight for the initial run.
Thanks.
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What sample rate is everyone using? Going to hook it up the the house board tonight for the initial run.
Thanks.
As a general practice, I record 96khz but usually master at 48
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I always record at 32/48.
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32/48
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I always record at 24/48.
On a 32 bit recorder? One with no input level control too? Curious?
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^ Given dallman's previous posts on the forum, I am pretty sure that was a typo. He is a big fan of 32-bit recording...
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Just had Nick make me a set if cables for this to go with my NBox. Planning to try the F3 out when I get home
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I always record at 32/48.
On a 32 bit recorder? One with no input level control too? Curious?
oops... I did mean 32/48, now corrected. :bigsmile:
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Curious, Why not run straight into the F3 for a smaller foot print?
Just had Nick make me a set if cables for this to go with my NBox. Planning to try the F3 out when I get home
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Curious, Why not run straight into the F3 for a smaller foot print?
Just had Nick make me a set if cables for this to go with my NBox. Planning to try the F3 out when I get home
The F3 has very good preamps. There is no real need for external preamps.
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Comp time!
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Comp time!
Agreed! Let the ABX-ing commence!
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I’ve ran a F3 once at an outdoor venue.. right next to the SBD. So it wasn’t necessarily perfect conditions, but decent.
Mikes > F3.
The recording sounds good.. but IMO.. it would have sounded better with a good external preamp.
But…... it was really slick to not have to worry about levels. YMMV.
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Some folks just like their preamps. Also, taking phantom power off the load on the batteries will significantly increase runtime.
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In addition to the nBox "flavor", another reason might be cabling. I guess it is probably easier/cheaper to get a short connect from the nBox to the recorder than to get new actives to go directly into the F3.
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Some folks just like their preamps. Also, taking phantom power off the load on the batteries will significantly increase runtime.
Definitely true, but being able to power the F3 via usb-c also makes for pretty much endless runtime (even with a pretty small battery block).
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Trying to figure this thing out before running it for the first time in a couple of weeks.
In the lower left of both waveforms there is a "x1 / x2 /x4" ...etc. This appears to be the gain. I know levels on this thing don't matter that much, but want to confirm what the best setting is for a normal concert.
Won't be my main rig, so no biggie if I screw it up, but might as well make a decent recording. :D
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Trying to figure this thing out before running it for the first time in a couple of weeks.
In the lower left of both waveforms there is a "x1 / x2 /x4" ...etc. This appears to be the gain. I know levels on this thing don't matter that much, but want to confirm what the best setting is for a normal concert.
Won't be my main rig, so no biggie if I screw it up, but might as well make a decent recording. :D
when I first got the F3 I left it at x1 and just raised levels in post sometimes as much as 20dB
I now go with x2 and it seems I only raise levels 5-10dB
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just hit record and put it on lock if you want, the slide. The waveforms are only visualizations I think. Its called magnification so that you can see a signal. It will set the levels by itself. someone correct me if I am wrong please.
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just hit record and put it on lock if you want, the slide. The waveforms are only visualizations I think. Its called magnification so that you can see a signal. It will set the levels by itself. someone correct me if I am wrong please.
NO... you have control of waveform BEFORE hitting record by using the menu
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The user manual doesn't really make it clear, but Darby is correct. You are in fact adjusting the recorded level. The good news is that you really can't mess up your levels recording with this kind of system (so long as you don't overload the input stage, which is unlikely).
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just hit record and put it on lock if you want, the slide. The waveforms are only visualizations I think. Its called magnification so that you can see a signal. It will set the levels by itself. someone correct me if I am wrong please.
NO... you have control of waveform BEFORE hitting record by using the menu
My mistake, thanks for the clarification. I keep mine set pretty high so that I can see it in the field. My files are always with some red and need normalization. This video spells it out better than the manual:
https://www.youtube.com/watch?source_ve_path=Mjg2NjQsMTY0NTAz&feature=emb_share&v=Jh34nsTETBI
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The user manual doesn't really make it clear, but Darby is correct. You are in fact adjusting the recorded level. The good news is that you really can't mess up your levels recording with this kind of system (so long as you don't overload the input stage, which is unlikely).
I know this is splitting hairs and makes no difference, but the Zoom F3 sets the gain, you set the faders which is the finished product either up or down if needed. To quote the speaker in the Zoom video, "I'll use the waveform magnifier as a fader to adjust the level of audio in headphones and on the recorded file"
The words "fader" and "recorded file" at least imply that the magnifier is working on a file after recording. Again, it really does not matter in the 32 bit float world, but I'm just relaying what the speaker is saying. And I know Sound Devices does let you set gain, but the Zoom 2, 3 and 6 (maybe Zoom 8xx, I do not have that deck) set gain and you adjust the faders. You are adjusting the (already) recorded file.
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The user manual doesn't really make it clear, but Darby is correct. You are in fact adjusting the recorded level. The good news is that you really can't mess up your levels recording with this kind of system (so long as you don't overload the input stage, which is unlikely).
I know this is splitting hairs and makes no difference, but the Zoom F3 sets the gain, you set the faders which is the finished product either up or down if needed. To quote the speaker in the Zoom video, "I'll use the waveform magnifier as a fader to adjust the level of audio in headphones and on the recorded file"
The words "fader" and "recorded file" at least imply that the magnifier is working on a file after recording. Again, it really does not matter in the 32 bit float world, but I'm just relaying what the speaker is saying. And I know Sound Devices does let you set gain, but the Zoom 2, 3 and 6 (maybe Zoom 8xx, I do not have that deck) set gain and you adjust the faders. You are adjusting the (already) recorded file.
I chose my words carefully when I said "recorded level" because the preamp gain is indeed fixed on the new F series if recording in 32-bit float mode (the only option on the F3).
The portion of the Zoom video you quote doesn't make it any clearer than the manual, unfortunately. No levels are being adjusted after it is recorded. What the "waveform adjustment" AKA faders are actually doing is adjusting the level post-ADC, and then that adjusted data is written to the file. See the last page of the F3 manual where the block diagram indicates exactly where that adjustment is happening.
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At risk of (a) getting it wrong or (b) confusing the issue further, it seems to me that the F3 preamps can't be adjusted by the user. What can be adjusted by the user is the position of the comparitively narrow ribbon of subsequently digitised audio within the wide ribbon of 32 bit float data.
I imagine seeing a DAW track, whose height represents the 1500dB (?) of 32 bit float data. The audio data would be seen as a quite thin waveform ribbon running along that DAW track.
In the F3, when recording the user can choose whether to position that thin waveform ribbon higher or lower within the imaginary broad DAW track height. You can't change how wide the ribbon of recorded data is within the DAW track, only how high or low within the track it is placed. But at the end of the day, it doesn't matter where it's positioned - it will sound the same after the D/A stage, if appropriate analog replay gain or attenuation is applied (so you don't explode your speakers, or have to press the speakers to your ears).
If someone tells me that's completely wrong I will happily delete this post!
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Looking for advice on the phone app. I have had it crap out on me twice. No biggie considering levels don't matter but what I am wondering is if the hold is not engaged on the recorder but only the phone app, will the hold still apply? I am not going to experiment during the show. It all seems to run fine despite the app losing connectivity and also wondering if reconnecting will blow the recording.
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I just took my F3 for it's maiden voyage a few nights ago. I appreciate everyone's input on the confusing "magnification settings". The fact of the matter is 32FP is still very new, a bit complicated and thus slightly confusing. Here are my initial thoughts:
-32FP IS AWESOME. IT IS WORTH THE INITIAL CONFUSION. I had a soundguy who was nice enough to offer use of his table right next to the soundboard. It was really crowded in the booth and with the crowd surrounding it so to be able to hit record, engage the hold button and just walk away was AWESOME. I didn't have to bother him or fend off other people to access the table and set my levels and so I wandered the venue for the entire show without worry. Reeaallly cool.
-It was a medium sized theater and a medium loud rock band. I decided to set the "magnification level" to x2 according to what I read here. Looking at the resulting wav file that I just transferred to my computer, I thought it was a bit low, BUT I ran my at853Rx's and I think the output of those is a bit low. Next time for a similar venue/band I will try x4 and report my findings. There is a free classical concert in the park this evening that I'd like to try my CM4's > F3 and I plan on trying magnification level x8. I will also report my findings if I'm able to get out there tonight.
-Thank you to Voltronic for looking at the actual manual and reporting the BLOCK DIAGRAM puts the magnification setting AFTER THE ADC's. I believe this confirms we are NOT affecting the actual gain stages before the (2) ADC's. WHATEVER the magnification settings are, we are not affecting the gain before the analog signal hits the ADC's.
-Now, the nerd/ scientist in me (and I'm guessing most of us) are wondering if the magnfication settings are affecting the 1's and 0's COMING OUT of the ADC. Is it the equivalent of (digital) gain being added to the file and thus changing the 1's and 0's? I honestly don't know and I still want to figure that out. But I think I can safely agree with what others have already said in that it's not a huge deal. I just performed the following test in my living room:
Using my F3, I recorded the exact same song in my living room but each time at different magnification settings. I started at x1, then x2, x4, x8, x16 etc etc all the way up to x1024! (it keeps doubling each time, for a total of 11 different magnification settings). After the 11th recording (at x1024) I cranked the volume of my stereo and recorded a 12th file (still at x1024) to really get those wav peaks flattened out to what we NORMALLY would consider clipping. I transferred all 12 different recordings of the exact same song, and placed them in Reaper. Below is a pic of the different wav files in a multi track setting. You can't see all 12 but you can clearly see how they actually get louder, confirming what others have said about it affecting how the final waveform APPEARS in a software setting.
Now listening to that 12th and final track, that APPEARS to be clipping, the track meters in Reaper are showing the red (actually pink) clipping indicators. Reaper was actually AUTOMUTING the track because it's technically so far over 0dB that it's engaging the AUTOMUTE feature to protect your speakers. What I could hear before the AUTOMUTE feature engaged did NOT sound like clipping. Then I opened that 12th track in Audacity and reduced the gain -20dB and reopened it in Reaper. It looked normal (see the 2nd pic below), and surprisingly sounded normal. There was ZERO distortion, clicks/pops or any negative audible artifacts that I could hear in playback. It is my initial belief this is because of the 1500dB (? plz correct) range of 32FP data (or gain range, plz correct) available that others have talked about. I still don't completely understand this but am simply accepting it for now, haha.
This tells me you cannot ruin a recording by mis-applying the magnification settings. With that being said, obviously don't test this by recording an important concert at x1024. I think we should all post our experiences with the magnification setting for the type of concert and how close it was to ideal "levels" and perhaps just accept the results haha. We'll then let the true audio engineers worry about the details. :)
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Update: I did make it to the free classical concert at the park and ran CM4's > F3 with a magnification level of x8. I just copied the files into Reaper and at x8 I still think they're too low. It is worth mentioning that I showed up late and ended up pretty far back by the "soundboard" and it was a good 80-100 ft back from the stage and it was pretty darn quiet. Shoutout to the old lady who finally turned around and shushed the volunteers who wouldn't stop yapping.
But while I was enjoying the concert, I was thinking about the magnification settings and what I believe is the truth: it still truly doesn't affect the final sound of your recording. No matter what, some type of gain is going to have to be applied. If I was using a normal 24-bit recorder, I would have applied analog gain to get the peaks around -10dB to -5dB. If I had set the magnification to x16 (which I would do if I could go back) then I believe Zoom's post-ADC digital gain (correct me if wrong) would have boosted the sound levels. But with my x8 recording, I ended up adding digital gain in Reaper to get the stereo file to an acceptable listening level.
I'm only thinking out loud here and welcome corrections, thoughts and feedback. If what I'm thinking is generally true, then we can at least have the argument about which gain is better: an analog preamp, Zoom's post-ADC digital gain (if that's what's happening, I'm still open to being wrong about this) or your DAW software gain/normalization. I can't help but remember reading a really old TS post about preamp gain affecting "soundstage".
Ultimately I hope everyone goes out and tapes more using these super cool devices and reports back. Comps are even better. If anyone in Denver with CM4's wants to comp an F3 and an Oade Warm Mod 661, that could be useful.
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...
Now listening to that 12th and final track, that APPEARS to be clipping, the track meters in Reaper are showing the red (actually pink) clipping indicators. Reaper was actually AUTOMUTING the track because it's technically so far over 0dB that it's engaging the AUTOMUTE feature to protect your speakers. What I could hear before the AUTOMUTE feature engaged did NOT sound like clipping. Then I opened that 12th track in Audacity and reduced the gain -20dB and reopened it in Reaper. It looked normal (see the 2nd pic below), and surprisingly sounded normal....
No need to go into Audacity. In REAPER without playing the track, hit F2 (Media Item Properties) and adjust the level. You can see the change in the track levels each time you hit Apply.
Fran
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No need to go into Audacity. In REAPER without playing the track, hit F2 (Media Item Properties) and adjust the level. You can see the change in the track levels each time you hit Apply.
Fran
ahhh nice 👍
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I'm only thinking out loud here and welcome corrections, thoughts and feedback. If what I'm thinking is generally true, then we can at least have the argument about which gain is better: an analog preamp, Zoom's post-ADC digital gain (if that's what's happening, I'm still open to being wrong about this) or your DAW software gain/normalization. I can't help but remember reading a really old TS post about preamp gain affecting "soundstage".
The primarily job of a preamp is amplification, and in that regard it is pretty safe to say you'll have no need to use one with the F3.
Technically, as with any analog circuit there are potential situations where the voltage levels of an input signal exceed the usable dynamic range of the F3's analog input stage - either so low level that the signal becomes swamped in the analog input stage's noise floor, or so high that it overloads it to the point of distortion or clipping. To make a good recording using the F3 in that kind of unusual situation (say recording the footsteps of small bugs, or a rocket launch at close proximity), you'd need to amplify or attenuate the signal ahead of the recorder, and could use an external preamp which exceeds the specs of the analog input stage of the F3 do that. Its just shifting signal level to comfortably fit within the real-world analog input dynamic range capability of the F3, which is quite large [edit- but minuscule in comparison to the ridiculously large dynamic range of the 32bit floating point format], large enough that you are unlikely to ever need to do that for concert recording.
Secondarily, some folks use a preamplifer to impart certain qualities to the sound that are felt to be desirable - a certain "flavor" or "color" that the "dry and clean" amplification input stage of the F-series recorders intentionally seeks to avoid. More subtly, one might perceive a different sense of "dimension" or "depth" or whatever, which might be described as presenting a different "soundstage". Using a preamp ahead of the F3 for these purpose will be common for some concert tapers.
Lastly, some microphones require special powering through a dedicated preamp, or have excessive phantom power requirements which might exceed what the recorder is able provide.
Once the signal has been digitized by the F3's ADC into the 32bit floating-point format, it shouldn't matter how you digitally adjust the signal level after that - which might be via F3 the magnification setting, or afterward in an digital editor capable of working with the 32bit-float format.
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With the Zoom M3 (the "MicTrak" device) all this is taken to another level of simplicity because there's no level setting of any kind (apart from headphone monitor volume control) nor even a display (so you don't need to concern yourself with "magnification" 'cos there's nothing to see being magnified). You just press the "record" button, and enjoy the show. This concept is made easier to achieve because the mic capsules are obviously chosen to suit the preamp - or the other way round - which removes any concerns about a mismatch in that respect.
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I do want to apologize for continuing to harp on a subject that has been covered ad nauseam (32FP). But I suppose it's relation to this specific recorder, the Zoom F3, and the confusion with it's "magnification" settings justifies the discussion.
If anyone is still considering an F3 or has one and is confused, don't worry. You can't mess anything up. Hit record, enjoy the show and this new technology. Your recording will be fine. :) I'm actually planning on taping a loud show with this set at the maximum x1024 magnfication setting to just prove this.
Secondarily, some folks use a preamplifer to impart certain qualities to the sound that are felt to be desirable - a certain "flavor" or "color" that the "dry and clean" amplification input stage of the F-series recorders intentionally seeks to avoid. More subtly, one might perceive a different sense of "dimension" or "depth" or whatever, which might be described as presenting a different "soundstage". Using a preamp ahead of the F3 for these purpose will be common for some concert tapers.
One thing I did not mention is how the F3 sounds. My initial impressions would agree with the "dry" and "clean" adjectives others have used to describe it. I would throw in "transparent" as well. The CM4's are pretty transparent and realistic mics and that super quiet outdoor classical concert I recorded sounded exactly how I remember it. I find the preamps to be very quiet; when I boosted the levels in my DAW software A LOT there was still no significant audible hiss. They are definitely quieter than the Tascam DR70D preamps.
I would recommend the F3 for sure, even if you're adverse to newer and confusing technologies. Like everyone says, it's smaller than it appears in pics. I got a tiny 5000mAh portable cell phone charger off Amazon and I'm pretty sure I would get festival run times. I only got 3 hours and 15 minutes off cheap Amazon Ni-mh AA rechargeables using my CM4's at 48v, but I got 6+ hours off the same rechargeable AA's powering my at853Rx's at 24v. I plan on running at853Rx > F3 as my go to on-the-fly rig and the smaller, super simple setup will get me taping more. The bag I've been using is so small I've been hanging it off the plastic knobs on my Manfrotto stand.
I wasn't even gonna get an F3 but some video guy on Facebook marketplace kept lowering the price until I couldn't say no. Glad I bought it!
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Just in case anyone is still confused:
1) The F3 (and also F6 and F8n PRO when set to 32-bit float mode) have a fixed input gain. That's because in this mode [Borg voice]: Gain is irrelevant. More accurately, it's set to a conservatively low level.
2) The multiple auto-ranging DACs take over from there, and are the special sauce that allows to you realize the benefits of the 32-bit FP data format the way it is being leveraged by Stagetec, Zoom, and Sound Devices.
3) Any adjustment of levels now is in the digital domain (and within the enormous dynamic range of the 32-bit FP data stream or container). That's why digital overs beyond 0 dBFS can be fully recovered.
4) Zoom's use of "magnification" terminology in regards to the F3 is confusing and/or misleading. You are adjusting the level being recorded, as a digital fader.
5) This tech still has limits. The analog input stage could be overloaded, and any clipping or distortion there is not recoverable. In practice, it is highly unlikely to happen unless recording extremely loud sources with extremely sensitive (hot output) mics. But input pads exist for a reason - you'll know if you need to use them.
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^ That all seems correct to me. A few comments-
Just in case anyone is still confused:
1) The F3 (and also F6 and F8n PRO when set to 32-bit float mode) have a fixed input gain. That's because in this mode [Borg voice]: Gain is irrelevant. More accurately, it's set to a conservatively low level.
2) The multiple auto-ranging DACs take over from there, and are the special sauce that allows to you realize the benefits of the 32-bit FP data format the way it is being leveraged by Stagetec, Zoom, and Sound Devices.
Don't want to take this thread down the rabbit hole, but the comment DSatz made over in the "Is levels peaking at 0 to high? (https://taperssection.com/index.php?topic=202622.0)" thread and my question about it posted earlier this morning (https://taperssection.com/index.php?topic=202622.msg2400229#msg2400229) is in currently in mind. To sum up- it may be more accurate to think of these systems as being auto-ranging combinations of multiple, specifically "tuned" input-gain-stages>ADCs, than as a single input stage feeding multiple auto-ranging ADCs. If so, this is something of an "ah-ha" moment for me in regards to understanding what is actually going on under the hood, but doesn't change how the recorders are used.
4) Zoom's use of "magnification" terminology in regards to the F3 is confusing and/or misleading. You are adjusting the level being recorded, as a digital fader.
I think "magnification" is less confusing than "level", but the "digital fader" concept is accurate. We are not adjusting level ahead of the ADC but after it, and in doing so are not affecting the accuracy of the data.. because (I think, the mathematics of this is where I'm weakest.. referring here to the Wikipedia entry on Floating-point arithmetic (https://en.wikipedia.org/wiki/Floating-point_arithmetic), which explains, In computing, floating-point arithmetic (FP) is arithmetic that represents subsets of real numbers using an integer with a fixed precision, called the significand, scaled by an integer exponent of a fixed base.) "magnification" is likely adjusting the floating-point exponent value, and not effecting the significand. In other words, "magnification" is scaling the output written to the file via the exponent but not otherwise changing what tapers call the "bit-accurate" output of the ADC, which is the significand.
To any mathematicians reading, please correct me if I'm off-base on this.
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Secondarily, some folks use a preamplifer to impart certain qualities to the sound that are felt to be desirable - a certain "flavor" or "color" that the "dry and clean" amplification input stage of the F-series recorders intentionally seeks to avoid. More subtly, one might perceive a different sense of "dimension" or "depth" or whatever, which might be described as presenting a different "soundstage". Using a preamp ahead of the F3 for these purpose will be common for some concert tapers.
I know it's more work, but if the F3's preamps are so 'transparent' (and can't be skipped anyway), shouldn't you be able to run the recording through a preamp of your choice afterwards and get the same results? If 'flavor' or 'color' is what you're after.
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Sure. Or emulate whatever particular sound you are listening for using plugins "in the box". I know some tapers who apply some light saturation in search of something like transformer or tape emulation. An advantage of good clean preamps is that you can always choose to alter things later when called for, or not, without that particular sound being permanently "baked in".
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Sure. Or emulate whatever particular sound you are listening for using plugins "in the box". I know some tapers who apply some light saturation in search of something like transformer or tape emulation. An advantage of good clean preamps is that you can always choose to alter things later when called for, or not, without that particular sound being permanently "baked in".
Add me to the list of people that prefer to record "clean" and add toppings later. The software is now good enough that you can get whatever flavor or adjustment you want. And can undo it.
Or if you prefer, you can still go old-school and do effect send/return through an analog mixer, if you have a great piece of gear that adds the flavor you want. But keep your unaltered original in case you change your mind down the road. I would try to keep AD/DA conversions to as few as possible, though.
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Sure. Or emulate whatever particular sound you are listening for using plugins "in the box". I know some tapers who apply some light saturation in search of something like transformer or tape emulation. An advantage of good clean preamps is that you can always choose to alter things later when called for, or not, without that particular sound being permanently "baked in".
Add me to the list of people that prefer to record "clean" and add toppings later. The software is now good enough that you can get whatever flavor or adjustment you want. And can undo it.
Or if you prefer, you can still go old-school and do effect send/return through an analog mixer, if you have a great piece of gear that adds the flavor you want. But keep your unaltered original in case you change your mind down the road. I would try to keep AD/DA conversions to as few as possible, though.
Agreed. Blood and guts can always be added later.
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Sure. Or emulate whatever particular sound you are listening for using plugins "in the box". I know some tapers who apply some light saturation in search of something like transformer or tape emulation. An advantage of good clean preamps is that you can always choose to alter things later when called for, or not, without that particular sound being permanently "baked in".
Any thoughts on the best settings to reproduce the flavor of some of the more popular external preamps we’re used to seeing? Would be cool to have something like an Oade M248 setting.
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Not specifically, although that's going to be the interpretive art of the undertaking.
I have thought about the possibility of doing what some of the more advanced guitar modeling amplifier folks are doing in "profiling" an amplifier or signal chain by sending test signals through that are in turn analyzed by the emulating unit to form a complex multi-dimensional profile of its signal-transfer-function. In essence sending complex signals through whatever is being tested, measuring its behavior across the entire frequency range, at all applicable levels, at various impedances, etc, and building a digital model of that behavior that can be applied to any signal passed in place of the real thing. At least that's my understanding of how those things work.
Profiling the transfer-function of a microphone preamp, a specific microphone itself, or a desired micrphone>preamp combination, would be a more fine-grained undertaking, as its effects are going to be much more subtle than the distortions of the cleanest guitar amplifier, yet is essentially no different in concept.
One example of this explained here-
https://youtu.be/N0aL-4UC60M?si=IDbtph3M4X4m1PlQ&t=84
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If you're looking for a colored mic preamp sound, you could go for one of the many Neve 1073 emulations, or look for plugins that do transformer or tube saturation.
Studio recordings are a completely separate thing, but if we're talking about concert recording:
My gut tells me that many people going after this drastic a change might be better served by being open to change up their mics and technique. If your recordings never sound "right" to you until you apply lots of coloration, are you truly capturing a concert event? Could you have gotten the "right" sound with different placement, height, spacing, or angles? Or different mics entirely? I'm not really including EQ or m/s processing into this, unless they are used with extreme settings.
We have to be open to new ways of doing things, and not be stuck in "this is THE way". I'm including myself in this advice, mind you. Several months ago, I used some very expensive omni mics to record a choral concert in a location and placement I had been successful with before, with different omni mics. As soon as I heard the recording, I knew I had chosen poorly. It was a mess; the direct/reverberant balance was all wrong. Next April I will be recording (and performing) in the same space, and I will not make that mistake again.
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I dream of a neutral preamp set with a pair of digital microphones with as little color as possible - Sonosax plus Neumann D? but that's off topic, sorry.
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Hi all. After wearing my old rig into the ground for 17+ years (DPA4061 legacy > MMA6000 or MPS6030 > solid state recorder), I've been researching a new one. The great beta in this thread convinced me that the F3 was a no-brainer. I paired it with a purchase of the newer DPA 4061 CORE mic kit with microdot > XLR adapters, and had a first run at a local jazz venue here in Seattle. Just wanted to drop some samples and observations here to add to the group data.
Samples -
https://drive.google.com/drive/folders/1OaZwZ5XP15lRsgdRCXs-GOwo7zjFjbrP?usp=sharing
Show: Oz Noy Trio Supergroup (Noy, Dennis Chambers, Jimmy Haslip) - 09/12/2023 - Seattle, WA @ Dimitriou's Jazz Alley.
Rig: DPA 4061 (CORE) mics > Zoom F3 via DPA DAD6001 XLR adapters. Mic in - 48v phantom power. 32 bit float / 48 kHz. Magnification x4.
Transfer: Zoom F3 > USB > PC > Reaper v6.82 DAW for normalizing and mastering > Render to 24/48 flac.
One sample is just normalized (+8db), no other edits. The other is the same song with some mastering in Ozone. The song (Bemsha Swing) has a very large dynamic range so you can hear the F3 preamps / noise floor in action.
Taping location was not ideal - right of stage without a direct line to a PA speaker, and it is bass guitar heavy being on Haslip's side. But I was close (5 foot off stage), so the sound was a mix between overhead PA speaker and the on-stage sounds of acoustic drums and monitors. I think the imperfections you hear (dampened sound, bass heavy, guitar distant) are a function of taping location rather than equipment.
Sound Quality Observations:
The F3 internal preamps sound clean, detailed and quiet to me. I was listening for F3 self-noise in the super quiet parts, but I don't hear much if any. I believe any light hum is from stage amps and snare. I'll need more shows for a definitive conclusion, but my first impression is that this is an upgrade from the old DPA MMA6000 preamp or MPS6030 battery box > Sony M10. The MMA6000 tended to add a touch of warm color that bordered on sloppy at times. The F3 is more neutral, detailed, and open. Could be also a function of the new core amps in the mics as well.
The 4061s > F3 seems like a natural fit, particularly if stealthing concerts is a primary purpose (like it is for me). I haven't yet seen this combo in shared shows on Dime and other sites.
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F3 Operational Observations:
- The 32 bit float is indeed a game-changer. No worrying about / checking levels?! I was able to stay more present in the performance.
- I set the magnification level to x4. That seemed to a good starting point for a club jazz show. Loud arena rock, maybe x2. Any others' thoughts on where to set it at a loud hard rock show? (keeping in mind it ultimately doesn't matter with the 32bit float. But the less heavy lifting my DAW editing software has to do, I think the better).
- I used Energizer lithium batteries and had PLENTY of juice left after recording for two hours (phantom 48v x 2 mics), then listening to the recording for another hour on the device. It hadn't even gone down a bar yet in the battery life indicator.
- I wore the F3 looped into my belt. Seems more natural than in the pocket, with it's thickness. But in a stealth situation it would be difficult to view the screen (upside down) and adjust anything via button pushing.
- After starting the recording, I slid the button to the Hold position and covered it with a small piece of gaffer's tape so it couldn't be knocked.
- The device automatically created a new track when the first track reached 1.99GB. This occurred about 1 hr 35mins into the show. It was seamless, no gap in music.
- 32 bit float worked as advertised in post editing. Increasing volume did not seem to introduce floor noise.
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Good report!
The song (Bemsha Swing) has a very large dynamic range so you can hear the F3 preamps / noise floor in action.
I'd be very, very, very, surprised if the noise floor of the recorder is what is dominating the noise floor of that recording. Granted, it can be hard to discern by listening if the noisefloor of the recording is being dominated by the noisefloor of the room (HVAC), the self-noise of the mics, or the noisefloor of the F3 input preamp stage. But from experience, I expect them to range from highest to lowest level in that order.
If you want to really test this you'd need to place a dummy load across the inputs of the F3 to hear just the noise of the recorder's input stage and determine its noise level. Then secondly, you could attach the mics and put the rig inside an airtight sealed case with lots of damping and make a recording like that, say, in the middle of the night at the quietest location you have available to you - maybe down in the basement, out in the back shed or in a well sealed car with the engine off and all the windows up. That's likely to get it quiet enough that the microphone self-noise will dominate, so you can then hear what that noise sounds like and determine its level, both in absolute terms and relative to the level of the recorder's input stage noise. Then you can compare both of those with the noise floor of this recording. You can then be sure its the room's noise floor if the noise level of the recording is higher than either of the other two.
Doing that will not only allow you to determine the true level of the noisefloor of each stage, but will also allow you to listen for differences in the spectrum and "flavor" of each stage's noise, which can help in determining what is dominating the noise floor of other recordings in the future by ear.
The MMA6000 tended to add a touch of warm color that bordered on sloppy at times. The F3 is more neutral, detailed, and open. Could be also a function of the new core amps in the mics as well.
In my experience, MMA6000 works well with the DPA miniatures, although it has significantly more gain available than necessary and is bulky. It isn't the quietest preamp, but works well at the gains typically needed for 4061 or 4060 where it is quieter than they are. When I shifted from MMA6000 to PFA's and DAD6001's I didn't notice much change into DR680mkII, noticed some change into OCM R-44, and noticed a lot more clarity and openness into Zoom F8.
I've been long curious about the CORE versions of the DPA miniatures. Comparison with the legacy version will be difficult here as you've got the additional variables of the DAD6001's and new recorder complicating that.
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Yes thanks for the report.
- I set the magnification level to x4. That seemed to a good starting point for a club jazz show. Loud arena rock, maybe x2. Any others' thoughts on where to set it at a loud hard rock show? (keeping in mind it ultimately doesn't matter with the 32bit float. But the less heavy lifting my DAW editing software has to do, I think the better).
One advantage of the magnification setting is eliminating the need to boost levels/ normalize in post. Now all I do after a show is copy the files, flac, then archive. Boom, done. This streamlined process gets me taping more.
I've even begun to run the magnification settings "hot" so that it appears there's some clipping (there's not, trust us) and then all my raw files are at consumer level playback volume. At my home theater I'm constantly switching between studio albums and live recordings and having everything at one general playback volume is quite nice. Not a HUGE deal, but still a pleasant convenience of the F3.
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Slighly off-topic. I always convert the orginal wav's to FLAC and make a md5 checksum file. Just for security and be able to check later if everyting is fine with files.
I do this with Traders Litle Helper. But it does nog work with 32bit files. Is there an alternative?
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Maybe try WavPack (https://www.wavpack.com)? FLAC can now do 32-bit, but only integer. I think they are working on implementing floating point.
Otherwise, just adjust the signal level, convert to 24-bit, and FLAC.
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I always convert the orginal wav's to FLAC and make a md5 checksum file. Just for security and be able to check later if everyting is fine with files.
You could md5 the raw 32-bit-fp files without conversion to FLAC, producing a checksum but not compressing file size.
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I always convert the orginal wav's to FLAC and make a md5 checksum file. Just for security and be able to check later if everyting is fine with files.
You could md5 the raw 32-bit-fp files without conversion to FLAC, producing a checksum but not compressing file size.
Really, man why did i not think of that....thanks mate!
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Happy to help!
..it can be hard to discern by listening if the noisefloor of the recording is being dominated by the noisefloor of the room (HVAC), the self-noise of the mics, or the noisefloor of the F3 input preamp stage. But from experience, I expect them to range from highest to lowest level in that order.
There should be sufficient data below to figure out the comparative noise floors of each. Specs as reported by the manufacturers:
Zoom's F3 EIN : -127 dBu or less (IHF-A) when waveform magnification is ×1024 with 150 Ω input (< not sure why the part I've bolded here is relevant)
Input impedance, MIC: 3 kΩ or more
Input impedance, LINE: 3 kΩ or more
Maximum input level, MIC: +4 dBu
Maximum input level, LINE: +24 dBu
DPA's MMA6000 EIN :
At 0 dB gain<3 µV (RMS A-weight)
At 37 dB gain<1,5 µV (RMS A-weight)
DPA's 4061 (legacy and CORE) :
Sensitivity, nominal, ±3 dB at 1 kHz 6 mV/Pa; -44 dB re. 1 V/Pa
Equivalent noise level, A-weighted Typ. 26 dB(A) re. 20 µPa (max. 28 dB(A))
Equivalent noise level, ITU-R BS.468-4 Typ. 38 dB (max. 40 dB)
Distortion, THD < 1% - Legacy 120 dB SPL RMS, 123 dB SPL peak
Distortion, THD < 1% - CORE 134 dB SPL RMS, 137 dB SPL peak
Dynamic range - Legacy Typ. 97 dB
Dynamic range - CORE Typ. 111 dB
Max. SPL, THD 10% 144 dB SPL peak
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I dream of a neutral preamp set with a pair of digital microphones with as little color as possible - Sonosax plus Neumann D? but that's off topic, sorry.
No analog preamp needed (or possible) with a digital microphone. Conversion from analog to digital is happening inside the microphone prior to output.
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Happy to help!
..it can be hard to discern by listening if the noisefloor of the recording is being dominated by the noisefloor of the room (HVAC), the self-noise of the mics, or the noisefloor of the F3 input preamp stage. But from experience, I expect them to range from highest to lowest level in that order.
There should be sufficient data below to figure out the comparative noise floors of each. Specs as reported by the manufacturers:
Zoom's F3 EIN : -127 dBu or less (IHF-A) when waveform magnification is ×1024 with 150 Ω input (< not sure why the part I've bolded here is relevant)
Input impedance, MIC: 3 kΩ or more
Input impedance, LINE: 3 kΩ or more
Maximum input level, MIC: +4 dBu
Maximum input level, LINE: +24 dBu
DPA's MMA6000 EIN :
At 0 dB gain<3 µV (RMS A-weight)
At 37 dB gain<1,5 µV (RMS A-weight)
DPA's 4061 (legacy and CORE) :
Sensitivity, nominal, ±3 dB at 1 kHz 6 mV/Pa; -44 dB re. 1 V/Pa
Equivalent noise level, A-weighted Typ. 26 dB(A) re. 20 µPa (max. 28 dB(A))
Equivalent noise level, ITU-R BS.468-4 Typ. 38 dB (max. 40 dB)
Distortion, THD < 1% - Legacy 120 dB SPL RMS, 123 dB SPL peak
Distortion, THD < 1% - CORE 134 dB SPL RMS, 137 dB SPL peak
Dynamic range - Legacy Typ. 97 dB
Dynamic range - CORE Typ. 111 dB
Max. SPL, THD 10% 144 dB SPL peak
I think you forgot to include the EIN specs of the MMA6000.
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I dream of a neutral preamp set with a pair of digital microphones with as little color as possible - Sonosax plus Neumann D? but that's off topic, sorry.
No analog preamp needed (or possible) with a digital microphone. Conversion from analog to digital is happening inside the microphone prior to output.
I always wondered why AES42 never became widely used. Only a few mic and field recorders still have models that use it.
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Happy to help!
..it can be hard to discern by listening if the noisefloor of the recording is being dominated by the noisefloor of the room (HVAC), the self-noise of the mics, or the noisefloor of the F3 input preamp stage. But from experience, I expect them to range from highest to lowest level in that order.
There should be sufficient data below to figure out the comparative noise floors of each. Specs as reported by the manufacturers:
Zoom's F3 EIN : -127 dBu or less (IHF-A) when waveform magnification is ×1024 with 150 Ω input (< not sure why the part I've bolded here is relevant)
Input impedance, MIC: 3 kΩ or more
Input impedance, LINE: 3 kΩ or more
Maximum input level, MIC: +4 dBu
Maximum input level, LINE: +24 dBu
DPA's MMA6000 EIN :
At 0 dB gain<3 µV (RMS A-weight)
At 37 dB gain<1,5 µV (RMS A-weight)
DPA's 4061 (legacy and CORE) :
Sensitivity, nominal, ±3 dB at 1 kHz 6 mV/Pa; -44 dB re. 1 V/Pa
Equivalent noise level, A-weighted Typ. 26 dB(A) re. 20 µPa (max. 28 dB(A))
Equivalent noise level, ITU-R BS.468-4 Typ. 38 dB (max. 40 dB)
Distortion, THD < 1% - Legacy 120 dB SPL RMS, 123 dB SPL peak
Distortion, THD < 1% - CORE 134 dB SPL RMS, 137 dB SPL peak
Dynamic range - Legacy Typ. 97 dB
Dynamic range - CORE Typ. 111 dB
Max. SPL, THD 10% 144 dB SPL peak
I think you forgot to include the EIN specs of the MMA6000.
It's there [I've now bolded it in the quote above], but is in µV rather than dBu.. and A-weighted.