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Gear / Technical Help => Recording Gear => Topic started by: Niels on September 06, 2024, 11:41:04 AM
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(http://)From Tascam’s IG account:
tascam_official Something big is coming for videographers, field recordists, and content creators...
• 2 XLR/TRS Combo jacks
• 32-bit float recording up to 192kHz
• Compact sturdy body that fits in videography rig
• USB-C Audio Interface
#TASCAM #videographer #fieldrecorder #contentcreator
Any guesses, insights, rumors??
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Fingers crossed they kept a DIGI-IN option......not crossing them too hard though :'(
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Seems to be the Zoom H1 XLR. Except made by Tascam. And probably with some more pro specs and price to match.
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Surely this is Tascam’s answer to the Zoom F3. It’ll be interesting if they can find a way to do a one-up. Smaller size? Better performance? -Mid-Side decoding?
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The timing indicates a product that could appeal to the same customers who would be in the market for an H1 XLR.
It is business 101 to pre-announce a product in your pipeline if it can possibly delay the user decision on a similar competitor’s product that may hit the market earlier than yours.
I notice there is no announcement/release date in the IG post above.
I am just hoping for “F3” preamp quality level and a 3.5mm stereo jack option. Wouldn’t mind some of the H1 XLR’s “mixing” options.
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I went to Tascam's IG page to check out the post and someone in the comments asked about a release date to which Tascam actually replied "by the end of the month".
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A Portacapture X2 or X4?
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DR100mk4 :smash:
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DR100mk4 :smash:
Wouldn't mind that, although the DR100 mk3 would hardly qualify as compact.
I think perhaps Tascam regards the X6 is the replacement of the DR100Mk3.
However hope this one is simpler than the X6/X8 - at least if I am to consider it.
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What's the chances of it having a touchscreen, like the X6/X8 recorders?
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What's the chances of it having a touchscreen, like the X6/X8 recorders?
I guess we'll find out at the end of the month
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Another teaser on IG:
An all-in-one solution for videographers
Generate, sync/receive time code
Jamsync ready for other TC devices
Allows wireless headphones
Control/Monitor up to five units.
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Ok... thanks for the update. Clearly not intended for same market as the Zoom H1 XLR, much more pro features for those needing them.
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It’s seeping through.
TASCAM FR-AV2
$399
https://www.bhphotovideo.com/c/product/1852610-REG/tascam_fr_av2_32_bit_float_recorder_timecode_generator.html
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A placeholder awaiting pictures and specs it seems.
But now we have a likely name and a price.
A Google search on the name didn't come up with anything else for me.
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It seems it doesn’t come with BT functionality built in.
$59
For the module.
https://www.bhphotovideo.com/c/product/1852609-REG/tascam_ak_bt2_bluetooth_adapter_for_fr_av2.html
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It seems it doesn’t come with BT functionality built in.
So a similar strategy in that aspect as the Zoom F3...
I am really curious about the size and form factor and if it would be any good for >:D usage. It clearly isn't the intended market of this device, so it seems there's still no affordable really small 32-bit float multi-adc stereo recorder yet...
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It seems it doesn’t come with BT functionality built in.
So a similar strategy in that aspect as the Zoom F3...
I am really curious about the size and form factor and if it would be any good for >:D usage. It clearly isn't the intended market of this device, so it seems there's still no affordable really small 32-bit float multi-adc stereo recorder yet...
Zoom H1 Essential
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Zoom H1 Essential
I'm not very fond of the Essential's noisy preamps, 'cause I will be taping stuff with high dynamics that need dynamic processing to make it a pleasant listening experience. I don't want relatively silent parts to get too noisy after amplification. H1XLR will likely be a bit better than H1 Essential, F3 is certainly better but clumsy shape for >:D. New Tascam input quality/shape/size unknown yet.
My ideal recorder seems to be an F2 form factor with F3-quality input and stereo 32-bit float... One should keep on dreaming, right?
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Zoom H1 Essential
I'm not very fond of the Essential's noisy preamps, 'cause I will be taping stuff with high dynamics that need dynamic processing to make it a pleasant listening experience. I don't want relatively silent parts to get too noisy after amplification. H1XLR will likely be a bit better than H1 Essential, F3 is certainly better but clumsy shape for >:D. New Tascam input quality/shape/size unknown yet.
My ideal recorder seems to be an F2 form factor with F3-quality input and stereo 32-bit float... One should keep on dreaming, right?
Do you find the F3 clumsy for stealth? I find my CA-11s > Zoom F3 rig less conspicuous than CA-11s (on croakies) > preamp (with a belt clip) > Zoom H1 (in my front pocket). I just turn the F3 on and put it in my back pocket.
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Do you find the F3 clumsy for stealth? I find my CA-11s > Zoom F3 rig less conspicuous than CA-11s (on croakies) > preamp (with a belt clip) > Zoom H1 (in my front pocket). I just turn the F3 on and put it in my back pocket.
Maybe my problem is that I just fairly recently started looking for a replacement recorder and found the F2 and the (then upcoming) Deity PR-2 (when everybody still expected it to be stereo). Both of them seem to be sized perfectly tiny but have other issues that disqualifies them for me (mono). Quality-wise the F3 seems perfect. But I don't need XLR and certainly compared to F2 and PR-2 it's oddly formed. I must confess: I've never hold an F3 in my hands yet. What I did do is building a Lego block with the size of an F3 and it made me realize I very likely won't be able to get it into stricter venues the way I usually do. But then again: I also realize that security seems be less strict than in the past, so it will likely still be possible to get it in anyway.
I guess my current line of candidates is:
1. F3 (despite it's form factor)
2. H1XLR (for lower price and hopefully good enough quality, pending proper reviews)
A new candidate could be the upcoming Tascam FR-AV2, pending combination of specs, size and prize. Not sure where that will land in my list of candidates.
The good thing is that I'm not in a hurry. I don't tape that often and my next gig is either in February or March next year...
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Do you find the F3 clumsy for stealth? I find my CA-11s > Zoom F3 rig less conspicuous than CA-11s (on croakies) > preamp (with a belt clip) > Zoom H1 (in my front pocket). I just turn the F3 on and put it in my back pocket.
Maybe my problem is that I just fairly recently started looking for a replacement recorder and found the F2 and the (then upcoming) Deity PR-2 (when everybody still expected it to be stereo). Both of them seem to be sized perfectly tiny but have other issues that disqualifies them for me (mono). Quality-wise the F3 seems perfect. But I don't need XLR and certainly compared to F2 and PR-2 it's oddly formed. I must confess: I've never hold an F3 in my hands yet. What I did do is building a Lego block with the size of an F3 and it made me realize I very likely won't be able to get it into stricter venues the way I usually do. But then again: I also realize that security seems be less strict than in the past, so it will likely still be possible to get it in anyway.
I guess my current line of candidates is:
1. F3 (despite it's form factor)
2. H1XLR (for lower price and hopefully good enough quality, pending proper reviews)
A new candidate could be the upcoming Tascam FR-AV2, pending combination of specs, size and prize. Not sure where that will land in my list of candidates.
The good thing is that I'm not in a hurry. I don't tape that often and my next gig is either in February or March next year...
In the UK, at clubs and academy-type venues, I've always placed my recorder + preamp in my backpack's smaller front pocket (the mics go in my spare glasses case). When it's my turn to be searched, I walk up to security with the main backpack compartment wide open - and they are always happy to take a look and leave it at that. No one's ever asked to search the smaller pockets.
At theatres, you don't even get searched most of the time.
I've never taped any arena or stadium shows in the UK, but I've brought stealth gear to festivals twice. The first time, they only searched the large pocket. The second time, they were a bit more thorough, but they failed to spot the F3 behind a sandwich.
I've only been to two shows in the US, both in 2016: Pitchfork Festival in Chicago, and Sufjan Stevens at Red Rocks. The guy at Pitchfork was very thorough and found my pre-amp and zoom H1 (he didn't care and let me in). I was not searched at Red Rocks.
Tldr; I've only ever had issues with cameras before. No one really cares about audio gear (though again, I've never tried taping in arenas or stadiums).
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In the UK...
Thanks a lot Rairun for describing your ways and the current security status in the UK. It doesn't seem to be very different from the status in my part of the world. It does make me feel more confident regarding getting the F3.
As my mics have a battery box and end in 3.5mm mini-jack, and I'm not in a hurry, I think I will wait regardless for what the H1XLR and FR-AV2 will bring. As the F3 doesn't have mini-jack input, I would have to get/make stereo mini-jack to XLR converters. (I won't need phantom-to-pip conversion due to the battery box. Probably a nice question for the 'cables section' when time is right.) These converters won't be necessary with the H1XLR as it already has 3.5mm input. We don't know yet about the FR-AV2...
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Do you find the F3 clumsy for stealth? I find my CA-11s > Zoom F3 rig less conspicuous than CA-11s (on croakies) > preamp (with a belt clip) > Zoom H1 (in my front pocket). I just turn the F3 on and put it in my back pocket.
Apologies if this is getting too off topic, but what do you use in between the CA-11s (which have a 1/8" plug, right?) and the F3? Do you have a preferred 1/8" > XLR adapter? Thanks!
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BACK ON TOPIC
anyone care to translate...
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/
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Do you find the F3 clumsy for stealth? I find my CA-11s > Zoom F3 rig less conspicuous than CA-11s (on croakies) > preamp (with a belt clip) > Zoom H1 (in my front pocket). I just turn the F3 on and put it in my back pocket.
Apologies if this is getting too off topic, but what do you use in between the CA-11s (which have a 1/8" plug, right?) and the F3? Do you have a preferred 1/8" > XLR adapter? Thanks!
I use these: https://store.lom.audio/products/usi-phantom-adapter?variant=4542168629280
They convert 24-48V phantom power into 8V PiP. I set the F3 to 24V, and that gives me 6 or 7h of battery life.
Thanks a lot Rairun for describing your ways and the current security status in the UK. It doesn't seem to be very different from the status in my part of the world. It does make me feel more confident regarding getting the F3.
As my mics have a battery box and end in 3.5mm mini-jack, and I'm not in a hurry, I think I will wait regardless for what the H1XLR and FR-AV2 will bring. As the F3 doesn't have mini-jack input, I would have to get/make stereo mini-jack to XLR converters. (I won't need phantom-to-pip conversion due to the battery box. Probably a nice question for the 'cables section' when time is right.) These converters won't be necessary with the H1XLR as it already has 3.5mm input. We don't know yet about the FR-AV2...
No worries! I wrote the post because I had many of the same worries as you before buying the F3 - not so much about being able to stealth successfully, but about how much more cumbersome the rig would be. I didn't feel the F3's size was a problem, but I did worry about the converters protruding out of it, so I held out for the Deity for a long time.
When the Deity turned out not to be a sound quality upgrade, I ended up just ordering the F3, and I don't regret it. I do think it would be cumbersome to have a set of adapters + a battery box, though - I mean, realistically it would be fine, I could make do, but what I like the most about my rig is how simple it is compared to using a pre-amp/battery box in the middle. Because of the locking XLR connector, for practical purposes, the adapter becomes part of the F3 (no danger of pulling at it and unplugging it). So you have a unit that you can stuff in your back pocket, and a single point of failure.
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BACK ON TOPIC
anyone care to translate...
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/
The specs are in English!
It looks very promising, tbh. If the pre-amp turns out to be as quiet as the F3's (and it appears to be: EIN of 127 dBu), I would choose it over the F3. The 3.5mm input provides up to 5V, so you wouldn't need a battery box nor a phantom adapter for a lot of small mics.
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It looks very promising, tbh. If the pre-amp turns out to be as quiet as the F3's (and it appears to be: EIN of 127 dBu), I would choose it over the F3. The 3.5mm input provides up to 5V, so you wouldn't need a battery box nor a phantom adapter for a lot of small mics.
:yahoo: Very promissing indeed! And despite the 3 AA batteries, the size seems a bit smaller than F3 (with the metal legs around it).
Some remarkable things:
XLR: +4 dBu max
TRS: +24 dBu max
So no switch or setting to choose line or mic in but hard-wired.
And there seems to be 'gain adjustment range' of 50dB. I assume that will be for 24bit only...
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BACK ON TOPIC
anyone care to translate...
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/
The specs are in English!
It looks very promising, tbh. If the pre-amp turns out to be as quiet as the F3's (and it appears to be: EIN of 127 dBu), I would choose it over the F3. The 3.5mm input provides up to 5V, so you wouldn't need a battery box nor a phantom adapter for a lot of small mics.
Professional 2-channel audio recorder with dual AD converters, sampling rates up to 192 kHz, resolution up to 32-bit float
Receives and sends timecode (LTC) for synchronisation with a video camera, simplifies post-production (Tentacle-ready)
Wireless timecode synchronisation possible with Atomos products*
Bluetooth audio monitoring possible with optional Tascam AK-BT2
Tascam Recorder Connect app allows integrated management of multiple FR-AV2 and DR-10L Pro* units
Two XLR/TRS microphone/line combo inputs (48 V or 24 V phantom power)
Ultra-compact user interface with 2-inch (5-cm) colour LCD
Other features
Ultra low-noise Tascam Ultra HDDA microphone preamplifier
High-precision clock generator (TCXO – Temperature Compensated Chrystal Oscillator)
Additional mini-jack input for audio from camera, wireless microphone receiver or plug-in powered microphones (2.5 V or 5 V)
Mini-jack output with adjustable attenuation allows audio to be sent to cameras and other devices
Switchable low-cut filter, compressor, limiter and noise gate for the inputs
3.5-mm stereo headphones jack
Built-in 2×2 USB audio interface, supports 32-bit float resolution
Adjustable input delay to compensate for distance between microphones (0–300 ms)
Adjustable output delay to compensate for time lag between camera video and audio output (0–300 ms)
Compact, lightweight and robust fiberglass enclosure designed for use with camera rigs
Various mark and auto-slate tone functions
Supports microSD/microSDHC/microSDXC recording media up to 512 Gigabytes
Metadata can be recorded (BEXT and iXML for WAV files)
Long-term recording with 3 AA batteries
Power can also be supplied via USB-C port
Specifications
Recorder specifications
Recording media
microSD (64 MB – 2 GB)
microSDHC (4–32 GB)
microSDXC (48–512 GB)
Recording/playback format
WAV (BWF)
Sampling frequency
48 kHz, 96 kHz, 192 kHz
Quantization bit depth
24 bit, 32 bit float
Input channels
2
Maximum number of recordable tracks
2
Analogue audio inputs and outputs
Microphone input (balanced)
XLR-3-31 (1: GND, 2: HOT, 3: COLD)
Maximum input level
+4 dBu
Minimum input level
–76 dBu
Gain adjustment range
0–50 dB
Input impedance
≥2 kΩ
Phantom power
24 V, 48 V
Line input (balanced)
6.3-mm standard TRS jacks
(Tip = HOT, Ring = COLD, Sleeve = GND)
Maximum input level
+24 dBu
Nominal input level
+4 dBu (GAIN at minimum)
Input impedance
≥8 kΩ
Line/timecode input (unbalanced)
3.5-mm stereo mini jack
Maximum input level
1 dBV
Nominal input level
–19 dBV (GAIN at minimum)
Minimum input level
–79 dBV
Input Impedance
≥6 kΩ (plug-in power off)
≥1.6 kΩ (plug-in power on)
Plug-in power
2.5 V, 5 V
Analogue audio outputs
Line/timecode output (unbalanced)
3.5-mm stereo mini jack
Maximum output level
+6 dBV
Nominal output level
–14 dBV
Output impedance
210 Ω
Headphones output
3.5-mm stereo mini jack
Maximum output power
50 mW + 50 mW (THD+N <0.1 % at 32 Ω)
Recommended headphones impedance
16–400 Ω
USB
Connector
USB Type-C
Transfer format
USB 2.0 HIGH SPEED
Device Class
Mass Storage Class
USB Audio Class 2.0
USB Audio
Sampling frequency
48 kHz, 96 kHz
Quantization bit depth
24 bit, 32 bit float
Number of inputs/outputs
2/2 channels
Timecode input/output
Timecode input
3.5-mm stereo mini jack
(Tip: Timecode IN, Ring: Not used, Sleeve: GND)
Input impedance
≥10 kΩ
Signal level range
0.5 to 5 Vpp
Timecode output
3.5-mm stereo mini jack
LTC selected
Terminal assignment
Tip: Timecode, Ring: Not used, Sleeve: GND
Output impedance
600 Ω
Output level
2 Vpp
MIC LEVEL selected
Terminal assignment
Tip: Timecode, Ring: right channel, Sleeve: GND
Output impedance
110 Ω
Output level
40 mVpp
Audio performance
EIN (equivalent input noise, mic preamp)
≤–127 dBu
Frequency response (input jacks to PCM data)
48 kHz: 20 Hz – 20 kHz, +0 dB / –0.5 dB
96 kHz: 20 Hz – 40 kHz, +0.5 dB / –1 dB
192 kHz: 20 Hz – 60 kHz, +0.5 dB / –3 dB
Dynamic range (input jacks to PCM data)
≥132 dB
(20-kHz SPCL LPF, A-weighting)
Total harmonic distortion ratio (THD+N, input jacks to PCM data)
≤0.02 %
(1 kHz sine wave, −1 dBFS input, GAIN at minimum, 20-kHz SPCL LPF)
Battery operation time
Conditions
Stereo recording at 48 kHz, both input jacks used, continuous operation, in hours:minutes (JEITA CP-2905)
Alkaline battery (Evolta)
09:00
Phantom power off, 24-bit resolution
04:30
Phantom power on (48 V, 2 × 3 mA), 32-bit float
NiMH battery (Eneloop)
08:30
Phantom power off, 24-bit resolution
05:30
Phantom power on (48 V, 2 × 3 mA), 32-bit float
Lithium battery (Energizer Ultimate Lithium)
15:00
Phantom power off, 24-bit resolution
10:00
Phantom power on (48 V, 2 × 3 mA), 32-bit float
Note: Operation times may vary depending on the microphones being used.
Power supply and other specifications
Power sources
3 AA batteries (alkaline, NiMH, lithium-ion)
USB bus power
TASCAM PS-P520U AC adapter (sold separately)
Power consumption
3.8 W (maximum)
Dimensions (W x H x D, overall)
99 mm × 80 mm × 40 mm
Weight
267 g (including batteries)
195 g (without batteries)
Operating temperature range
0–40 °C
Included items
Safety guide, Tascam ID registration guide
* Requires optional Bluetooth adapter AK-BT1 or AK-BT2
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Apologies if this is getting too off topic, but what do you use in between the CA-11s (which have a 1/8" plug, right?) and the F3? Do you have a preferred 1/8" > XLR adapter? Thanks!
I use these: https://store.lom.audio/products/usi-phantom-adapter?variant=4542168629280
They convert 24-48V phantom power into 8V PiP. I set the F3 to 24V, and that gives me 6 or 7h of battery life.
Thank you :coolguy:
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Thanks for the specs on the recorder. I'm hoping the lack of it mentioning touch screen means it won't have a touch screen and will be controlled by those F keys.
I'm guessing the bluetooth dongle will plug into the top of the unit, behind that plastic door.
It has USB-c but the doc says Transfer format: USB 2.0 HIGH SPEED.
Since this seems to be marketed towards videographers, field recordists, and content creators, I'm hoping that means adjustable gain/input on each channel, either on the unit or via bluetooth.
Maybe more specs and docs will be leaked or else we'll be waiting a little bit longer.
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It’s seeping through.
TASCAM FR-AV2
$399
https://www.bhphotovideo.com/c/product/1852610-REG/tascam_fr_av2_32_bit_float_recorder_timecode_generator.html
Looks like they took it down. Page not found today
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no digi ;(
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no digi ;(
Wouldn't using the deck as a bit bucket negate any benefit of using a 32 bit recorder? Digi in means it is storing the bits from another device. Just wondering if I am missing something obvious?
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If you have a dac that outputs 32bit you’d want a recorder to save that.
Dig in on a small form factor recorder is useful whether you want 24 or 32 bit files.
Also, 32 bit is mostly just for marketing purposes. 24 bit is more than enough headroom for anything you want to record, just takes setting your levels conservatively.
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no digi ;(
Wouldn't using the deck as a bit bucket negate any benefit of using a 32 bit recorder? Digi in means it is storing the bits from another device. Just wondering if I am missing something obvious?
If I'm running out from my minime, yes, I'd love to have a digi in and have a super small recorder (as a bit bucket, and just run 24/96), and if I have wanted to just run the V2 in front, then I could just line in. But yes, I'd love to have an option of a digi in being it spdif or aes (personally, of course! ;D )
the unit does look pretty nice though, regardless!
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This one?
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/ (https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/)
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This one?
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/ (https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/)
Yep! But be aware that the display in the photos must be a mock-up and not the real thing. It is unrealistically sharp and if it would be real, there must be an SD card of ~1.5TB in it to still have 1234 hours remaining, while it seems to support max 512GB...
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This one?
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/ (https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/)
SC Sound is the Danish distributor so the price displayed is likely the recommended retail price including Danish 25% VAT.
The current street price of the Zoom F3 is about 20% lower, so I would guess that the Tascam will end at the same price point as the Zoom F3 when it hits the street.
I don’t need all the TC stuff but other than that, it looks like this recorder will check most boxes on my list.
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No "hold" button is a deal breaker for me. Maybe it's engaged with a menu setting, but if not......
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Seems to be a bit larger and heavier than the F3
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Like the Portacapture X6/X8 recorders, the FR-AV2 doesn’t support phantom power in line-in mode. I really like having that ability (essentially a -20db preamp input pad) with my Zoom F3, for recording very loud sounds with my high output condenser mics. A +4dBu max input level before preamp input clipping can be pretty low headroom in some cases. For comparison, the Mixpre series recorder XLR inputs have a native +14dBu of headroom before input clipping in mic input mode.
In any case, I’ll be interested to see this new recorder in the wild, and hear real user experiences with it.
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M/S not mentioned... overall it is clearly a more upmarket device than the Zoom H1 XLR - but choices are good. All things considered this generation of device is stupidly cheap for what you get. Apart from bulk. You don't get much of a lump for your money. :bigsmile:
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no digi ;(
it looks like it might do digital audio via usb c, meaning something like an m2d2 could feed it a usb digital signal
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no digi ;(
it looks like it might do digital audio via usb c, meaning something like an m2d2 could feed it a usb digital signal
One side or the other has to be the host. Neither the recorder nor the sonosax box have that capability.
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In German
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/?fbclid=IwY2xjawFYU7VleHRuA2FlbQIxMAABHdubcXY83utPG9V2R-986n_TiM_M7ymCnFbLeNBuo_hRM5wYrBPmfEDfhg_aem_sxix85gH6dyeoaHg84zBPA
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Interesting, yet, only 48/96/192kHz
No 44.1?
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That's Danish. But close enough.
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In German
https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/?fbclid=IwY2xjawFYU7VleHRuA2FlbQIxMAABHdubcXY83utPG9V2R-986n_TiM_M7ymCnFbLeNBuo_hRM5wYrBPmfEDfhg_aem_sxix85gH6dyeoaHg84zBPA
I’m Danish, but take no offense. I get it; “some strange language”😆
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Adding these here as they have taken all the links down.
Professional 2-channel audio recorder with dual AD converters, sampling rates up to 192 kHz, resolution up to 32-bit float
Receives and sends timecode (LTC) for synchronisation with a video camera, simplifies post-production (Tentacle-ready)
Wireless timecode synchronisation possible with Atomos products*
Bluetooth audio monitoring possible with optional Tascam AK-BT2
Tascam Recorder Connect app allows integrated management of multiple FR-AV2 and DR-10L Pro* units
Two XLR/TRS microphone/line combo inputs (48 V or 24 V phantom power)
Ultra-compact user interface with 2-inch (5-cm) colour LCD
Other features
Ultra low-noise Tascam Ultra HDDA microphone preamplifier
High-precision clock generator (TCXO – Temperature Compensated Chrystal Oscillator)
Additional mini-jack input for audio from camera, wireless microphone receiver or plug-in powered microphones (2.5 V or 5 V)
Mini-jack output with adjustable attenuation allows audio to be sent to cameras and other devices
Switchable low-cut filter, compressor, limiter and noise gate for the inputs
3.5-mm stereo headphones jack
Built-in 2×2 USB audio interface, supports 32-bit float resolution
Adjustable input delay to compensate for distance between microphones (0–300 ms)
Adjustable output delay to compensate for time lag between camera video and audio output (0–300 ms)
Compact, lightweight and robust fiberglass enclosure designed for use with camera rigs
Various mark and auto-slate tone functions
Supports microSD/microSDHC/microSDXC recording media up to 512 Gigabytes
Metadata can be recorded (BEXT and iXML for WAV files)
Long-term recording with 3 AA batteries
Power can also be supplied via USB-C port
Specifications
Recorder specifications
Recording media
microSD (64 MB – 2 GB)
microSDHC (4–32 GB)
microSDXC (48–512 GB)
Recording/playback format
WAV (BWF)
Sampling frequency
48 kHz, 96 kHz, 192 kHz
Quantization bit depth
24 bit, 32 bit float
Input channels
2
Maximum number of recordable tracks
2
Analogue audio inputs and outputs
Microphone input (balanced)
XLR-3-31 (1: GND, 2: HOT, 3: COLD)
Maximum input level
+4 dBu
Minimum input level
–76 dBu
Gain adjustment range
0–50 dB
Input impedance
≥2 kΩ
Phantom power
24 V, 48 V
Line input (balanced)
6.3-mm standard TRS jacks
(Tip = HOT, Ring = COLD, Sleeve = GND)
Maximum input level
+24 dBu
Nominal input level
+4 dBu (GAIN at minimum)
Input impedance
≥8 kΩ
Line/timecode input (unbalanced)
3.5-mm stereo mini jack
Maximum input level
1 dBV
Nominal input level
–19 dBV (GAIN at minimum)
Minimum input level
–79 dBV
Input Impedance
≥6 kΩ (plug-in power off)
≥1.6 kΩ (plug-in power on)
Plug-in power
2.5 V, 5 V
Analogue audio outputs
Line/timecode output (unbalanced)
3.5-mm stereo mini jack
Maximum output level
+6 dBV
Nominal output level
–14 dBV
Output impedance
210 Ω
Headphones output
3.5-mm stereo mini jack
Maximum output power
50 mW + 50 mW (THD+N <0.1 % at 32 Ω)
Recommended headphones impedance
16–400 Ω
USB
Connector
USB Type-C
Transfer format
USB 2.0 HIGH SPEED
Device Class
Mass Storage Class
USB Audio Class 2.0
USB Audio
Sampling frequency
48 kHz, 96 kHz
Quantization bit depth
24 bit, 32 bit float
Number of inputs/outputs
2/2 channels
Timecode input/output
Timecode input
3.5-mm stereo mini jack
(Tip: Timecode IN, Ring: Not used, Sleeve: GND)
Input impedance
≥10 kΩ
Signal level range
0.5 to 5 Vpp
Timecode output
3.5-mm stereo mini jack
LTC selected
Terminal assignment
Tip: Timecode, Ring: Not used, Sleeve: GND
Output impedance
600 Ω
Output level
2 Vpp
MIC LEVEL selected
Terminal assignment
Tip: Timecode, Ring: right channel, Sleeve: GND
Output impedance
110 Ω
Output level
40 mVpp
Audio performance
EIN (equivalent input noise, mic preamp)
≤–127 dBu
Frequency response (input jacks to PCM data)
48 kHz: 20 Hz – 20 kHz, +0 dB / –0.5 dB
96 kHz: 20 Hz – 40 kHz, +0.5 dB / –1 dB
192 kHz: 20 Hz – 60 kHz, +0.5 dB / –3 dB
Dynamic range (input jacks to PCM data)
≥132 dB
(20-kHz SPCL LPF, A-weighting)
Total harmonic distortion ratio (THD+N, input jacks to PCM data)
≤0.02 %
(1 kHz sine wave, −1 dBFS input, GAIN at minimum, 20-kHz SPCL LPF)
Battery operation time
Conditions
Stereo recording at 48 kHz, both input jacks used, continuous operation, in hours:minutes (JEITA CP-2905)
Alkaline battery (Evolta)
09:00
Phantom power off, 24-bit resolution
04:30
Phantom power on (48 V, 2 × 3 mA), 32-bit float
NiMH battery (Eneloop)
08:30
Phantom power off, 24-bit resolution
05:30
Phantom power on (48 V, 2 × 3 mA), 32-bit float
Lithium battery (Energizer Ultimate Lithium)
15:00
Phantom power off, 24-bit resolution
10:00
Phantom power on (48 V, 2 × 3 mA), 32-bit float
Note: Operation times may vary depending on the microphones being used.
Power supply and other specifications
Power sources
3 AA batteries (alkaline, NiMH, lithium-ion)
USB bus power
TASCAM PS-P520U AC adapter (sold separately)
Power consumption
3.8 W (maximum)
Dimensions (W x H x D, overall)
99 mm × 80 mm × 40 mm
Weight
267 g (including batteries)
195 g (without batteries)
Operating temperature range
0–40 °C
Included items
Safety guide, Tascam ID registration guide
* Requires optional Bluetooth adapter AK-BT1 or AK-BT2
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DR-AV2 metering > F3's If it sounds anything like the DR-100 mk3, it's a homerun.
I'll gonna buy one. Original Tascam fanboy! lol
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Does it record to an SD card of some type? I am not sure I am seeing that in the specs.
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Does it record to an SD card of some type? I am not sure I am seeing that in the specs.
micro sdxc
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I am way more impressed than I expected to be. It appears thinner than the Zoom F3
From a >:D position if the 5V pip is true, that would be big.
3.5-mm stereo mini jack
Maximum input level
1 dBV
Nominal input level
–19 dBV (GAIN at minimum)
Minimum input level
–79 dBV
Input Impedance
≥6 kΩ (plug-in power off)
≥1.6 kΩ (plug-in power on)
Plug-in power
2.5 V, 5 V
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And hopefully and eventual "AV-4 or AV-6!"
-
I made a quick comparison of the physical dimensions of both before they took down the specs for the new Tascam. As I recall, the Tascam was slightly thinner then the F3 but it was also slightly larger in the other dimensions and heavier.
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I saved the danish dealer page to archive org's wayback machine.
https://web.archive.org/web/20240918144151/https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/ (https://web.archive.org/web/20240918144151/https://scsound.dk/shop/23-haandholdt-recorder/23248-tascam-2-kanals-audio-recorder/)
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And here’s the original page as pdf.
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A new post on their instagram page: https://imginn.com/p/DAJN6NsJq9H/
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I am misssing the hold function switch...I will not buy a recorder without it....
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I am misssing the hold function switch...I will not buy a recorder without it....
I can’t imagine there’s no locking mechanism at all. Hopefully it’ll be implemented in a practical way. Could it be that the on/off slider has a double function for this? Slide to one side (and keep it that way for a few seconds) is on/off, slide it the other way is lock?? I am curious too, time will tell…
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I am misssing the hold function switch...I will not buy a recorder without it....
I can’t imagine there’s no locking mechanism at all. Hopefully it’ll be implemented in a practical way. Could it be that the on/off slider has a double function for this? Slide to one side (and keep it that way for a few seconds) is on/off, slide it the other way is lock?? I am curious too, time will tell…
The tascam dr-10L has a slider switch for power and start/stop recording. When you're recording, you can't access the menu until you stop recording. The 10L is a tiny recorder meant to be bodyworn, so it makes sense the controls are locked.
With this AV2 recorder, i don't think it would make sense to start/stop recording to access the menu and make changes.
Maybe it'll be a slide up and hold kind of mechanism where it will lock controls out, and a slide up and hold to unlock.
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Manual at https://tascam.jp/int/product/fr-av2/support (https://tascam.jp/int/product/fr-av2/support) !
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Yes!
Recording hold function (REC HOLD)
This function is to prevent recording from being unintentionally stopped by misoperation. When this function is on, recording can be
stopped by pressing and holding the STOP button.
o The recording hold function is on when shipped new from the factory.
Set the recording hold function using the MENU screen > REC SETTINGS > REC HOLD item.
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It only seems to prevent stopping recording, nothing else... Not sure about other things that may accidentally go wrong during recording, need to read some more... ;D
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5V Plug in Power!
I’m getting my hopes up that this could be the 32-bit set and forget stealth recorder we have been hoping for.
The dimensions are 99 mm x 80.4 mm x 40.3 mm … that’s definitely small enough to sneak into venues I reckon.
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5V Plug in Power!
I’m getting my hopes up that this could be the 32-bit set and forget stealth recorder we have been hoping for.
The dimensions are 99 mm x 80.4 mm x 40.3 mm … that’s definitely small enough to sneak into venues I reckon.
From the manual:
Setting plug-in power (PLUG IN POWER)
Options: OFF (default), 2.5V, 5V
When connecting a microphone that requires plug-in power, set
this to “2.5V” or “5V”.
CAUTION
Do not turn plug-in power on when a mic is connected that
does not require it. Doing so could damage the connected
equipment.
-
Tascam official video.
https://www.youtube.com/watch?v=lCJ8CU6fqNs
B&H Photo demo
https://www.youtube.com/watch?v=f3oiBmdwT9Y
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"Tascam Official" liked this comments on their video:
"Don’t change anything and make a 4 XLR version for around $500 and I will ditch Zoom"
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no digi ;(
it looks like it might do digital audio via usb c, meaning something like an m2d2 could feed it a usb digital signal
One side or the other has to be the host. Neither the recorder nor the sonosax box have that capability.
I'm not sure this is true. Tascam says this thing has 2 way usb audio
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B&H Photo demo
https://www.youtube.com/watch?v=f3oiBmdwT9Y
I liked how the presenter switched from 24 bit mode to 32 bit without having to reboot the recorder. Sound Devices, take notice! ;)
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So it does have 24 bit recording and it looks like the levels can be adjusted while recording. Plus presets can be saved.
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no digi ;(
it looks like it might do digital audio via usb c, meaning something like an m2d2 could feed it a usb digital signal
One side or the other has to be the host. Neither the recorder nor the sonosax box have that capability.
I'm not sure this is true. Tascam says this thing has 2 way usb audio
If connected over usbc to a computer or phone with usbc for example, it acts like an audio io card with two channels in and two channels out.
USB Audio
Sampling frequency
48/96 kHz
Quantization bit depth: 24-bit/32-bit float
Number of input channels: 2 (output from unit)
Number of output channels: 2 (input to unit)
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Dang, this thing looks beautiful. Looks like it's set to be released today? Does that mean suppliers still need some time to get it in stock? Looks like it's still 'preorder' status on BHP. I think I will have to replace my A10s with this, I am hoping their Bluetooth adapters are a bit more reliable in large rooms. That plus 32bit float will eliminate the small bit of pain I still experience on a lot of my runs.
-
Bye bye Zoom F3 ;D
-
It also has dual bluetooth. It can pair with earbuds and an external speaker at the same time.
This thing is loaded with goodies! lol
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accessory related question, in the comments he says all his setup is on Amazon but he doesn't specify all the details.
His rig is really interesting, do you know what are the accessories he uses?
Thanks !
https://youtu.be/AQGc4Ui6rp8?si=VZ79WgTN8waunC-i&t=351
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accessory related question, in the comments he says all his setup is on Amazon but he doesn't specify all the details.
His rig is really interesting, do you know what are the accessories he uses?
Thanks !
https://youtu.be/AQGc4Ui6rp8?si=VZ79WgTN8waunC-i&t=351
It does look convenient for his use-case.
What exactly caught your interest?
He gives a few clues:
Microphones and shock mounts and other mounting options can be found on LOM audio.
An equivalent stereo bar can be had from many sources, just search "stereo bar" on amazon.
He say the tripod shown is from his DJI smartphone gimbal.
The handle could be any with top and button tripod screws. A search for "grip with 1/4 inch screw" would get you started.
I am not sure he himself know the brands of the small accessories, as it sounds like he enjoys finding those things cheaply on Amazon. And even if he knew, it may well be some Chinese off-brand that doesn't exist anymore.
He does mention that it comes from his video rig, and if you search for things like "dual cold shoe", "nato quick release", "cold shoe ball head" and brands like SmallRig, Nitze, Ulanzi, Neweer etc. you will find a lot of inexpensive video related mounting systems and accessories that can be used for creating recording rigs that can be easily dismantled and reassembled.
Video rigs are extremely modular and there are many vendors. Although intended for video, these components can be used for any application where tripod 1/4 and 3/8 screw mounts are commonly used and only your imagination defines the limits.
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With 32bit float, true TC support and audio monitoring of multiple units I am guessing this will be a big hit with videographers, especially the many with no or tiny crews.
Typically a videographer would want more than two channels, but with TC you can just get the number of FR-AV2 units needed to fulfil your channel requirement and place them on or near the sound source and easily sync with TC in post.
Critical gain monitoring is unnecessary with 32bit float, and you can still verify actual audio remotely with BT if required.
It certainly looks like it will fulfil most, if not all, my requirements. I personally wouldn't need TC for the type of video sound I shoot, but both as a field and live music recorder it seems to tick all the boxes relevant for me.
I wouldn't mind a normalisation function in case I need a quick 24bit output. I can't see that being mentioned so far, but maybe it could be included in a future firmware update.
I just need to tame my excitement and wait until the price settles 20-25% lower than introduction price - as Tascam recorders normally do :-)
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accessory related question, in the comments he says all his setup is on Amazon but he doesn't specify all the details.
His rig is really interesting, do you know what are the accessories he uses?
Thanks !
https://youtu.be/AQGc4Ui6rp8?si=VZ79WgTN8waunC-i&t=351
It does look convenient for his use-case.
What exactly caught your interest?
He gives a few clues:
Microphones and shock mounts and other mounting options can be found on LOM audio.
An equivalent stereo bar can be had from many sources, just search "stereo bar" on amazon.
He say the tripod shown is from his DJI smartphone gimbal.
The handle could be any with top and button tripod screws. A search for "grip with 1/4 inch screw" would get you started.
I am not sure he himself know the brands of the small accessories, as it sounds like he enjoys finding those things cheaply on Amazon. And even if he knew, it may well be some Chinese off-brand that doesn't exist anymore.
He does mention that it comes from his video rig, and if you search for things like "dual cold shoe", "nato quick release", "cold shoe ball head" and brands like SmallRig, Nitze, Ulanzi, Neweer etc. you will find a lot of inexpensive video related mounting systems and accessories that can be used for creating recording rigs that can be easily dismantled and reassembled.
Video rigs are extremely modular and there are many vendors. Although intended for video, these components can be used for any application where tripod 1/4 and 3/8 screw mounts are commonly used and only your imagination defines the limits.
Thanks a lot Niels, I'll have a search with all the details you give for the video accessories (beeing a non-English speaker doesn't help..)
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Thanks a lot Niels, I'll have a search with all the details you give for the video accessories (beeing a non-English speaker doesn't help..)
You are welcome. If your interest was caught by the quick release you should be aware that some of the video related systems may require matching systems to attach to each other. For example a Nato style quick release or clamp will require a Nato rail to be attached to or an ARRI style handle will require an ARRI compatible mount point.
-
Regarding hold function: https://youtu.be/aHPAmtBhWjo?si=LrCQwf-EV7WUcNGS&t=548
There's a menu option to prevent the recorder from being stopped during recording, but it does not lock out the F1-F4 keys.
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Bye bye Zoom F3 ;D
And I just bought an F3 not too long ago, would have held out for this if I had know about it. Not happy with no built in bluetooth (in either) but probably pick one up when the price drops a little.
-
Regarding hold function: https://youtu.be/aHPAmtBhWjo?si=LrCQwf-EV7WUcNGS&t=548
There's a menu option to prevent the recorder from being stopped during recording, but it does not lock out the F1-F4 keys.
If you look at the pictures on the Tascam site you can see that the keys for F1-F4 are recessed, so pushing those keys unintentionally is not likely.
-
Regarding hold function: https://youtu.be/aHPAmtBhWjo?si=LrCQwf-EV7WUcNGS&t=548
There's a menu option to prevent the recorder from being stopped during recording, but it does not lock out the F1-F4 keys.
He’s claiming just under 11 hours with lithium batteries.
I expect most people will be using rechargeable batteries with 2/3rds the capacity which equates to 7 hours.
What’s the betting that in reality it’s nearer to the Mixpre 3 battery life of 2.5 hours?
-
Regarding hold function: https://youtu.be/aHPAmtBhWjo?si=LrCQwf-EV7WUcNGS&t=548
There's a menu option to prevent the recorder from being stopped during recording, but it does not lock out the F1-F4 keys.
He’s claiming just under 11 hours with lithium batteries.
I expect most people will be using rechargeable batteries with 2/3rds the capacity which equates to 7 hours.
What’s the betting that in reality it’s nearer to the Mixpre 3 battery life of 2.5 hours?
Used phantom power on Zoom F3 for a show on Friday with lithiums. Still had full bars after 2.5 hours so I can believe it.
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What’s the betting that in reality it’s nearer to the Mixpre 3 battery life of 2.5 hours?
Power usage MixPre 3 is with phantom power switched on about 6Watt (not correct, its much higher). Maximum power usage F3 is 5watt. Maximum power usage Tascam FR-AV2 is 3.8Watt. The reason that the power usage of the Zoom F3 is higher has to do that it only has 2 AA batteries for power, ie maximum 3Volt. Tascam has 3AA for power, ie maximum 4.5Volt. To give you an idea what's happening, lets take the 48Volt for phantom power. The microphone uses 3mA with 48Volt. That is 0.144Watt. For the Zoom F3 this will be at least 0.144Watt / 3Volt = 48mA for a mic. For the Tascam FR-AV2 this will be at least 0.144Watt / 4.5Volt = 10,7mA. If you use two microphones this means for the Zoom 96mA and for the Tascam 21.4mA, difference 74.6mA.
-
Regarding hold function: https://youtu.be/aHPAmtBhWjo?si=LrCQwf-EV7WUcNGS&t=548
There's a menu option to prevent the recorder from being stopped during recording, but it does not lock out the F1-F4 keys.
He’s claiming just under 11 hours with lithium batteries.
I expect most people will be using rechargeable batteries with 2/3rds the capacity which equates to 7 hours.
What’s the betting that in reality it’s nearer to the Mixpre 3 battery life of 2.5 hours?
Used phantom power on Zoom F3 for a show on Friday with lithiums. Still had full bars after 2.5 hours so I can believe it.
Yup, Energizer Ultimate Lithiums gives me 6.5 hrs powering CCM 4 mics with the F3
-
From the manual:
BACKLIGHT
This sets the display backlight.
OFF: Backlight always off
5–30 sec: Backlight turns off automatically after set time without operation
ALWAYS: Backlight always stays on (default)
NOTE
The backlight setting is only active during battery operation.
The backlight will always stay lit when operating on USB bus
I just don't get it why Tascam wants to overrule the user preference when USB power is connected. Such a pity. Hopefully they can and will change this odd behavior with a firmware update.
-
From the manual:
BACKLIGHT
This sets the display backlight.
OFF: Backlight always off
5–30 sec: Backlight turns off automatically after set time without operation
ALWAYS: Backlight always stays on (default)
NOTE
The backlight setting is only active during battery operation.
The backlight will always stay lit when operating on USB bus
I just don't get it why Tascam wants to overrule the user preference when USB power is connected. Such a pity. Hopefully they can and will change this odd behavior with a firmware update.
I am not sure that necessary is the correct conclusion from what the manual is saying. "Operation on the USB bus" is not the same as "when USB power is connected" - they may be referring to audio interface usage.
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From the manual:
BACKLIGHT
This sets the display backlight.
OFF: Backlight always off
5–30 sec: Backlight turns off automatically after set time without operation
ALWAYS: Backlight always stays on (default)
NOTE
The backlight setting is only active during battery operation.
The backlight will always stay lit when operating on USB bus
I just don't get it why Tascam wants to overrule the user preference when USB power is connected. Such a pity. Hopefully they can and will change this odd behavior with a firmware update.
I am not sure that necessary is the correct conclusion from what the manual is saying. "Operation on the USB bus" is not the same as "when USB power is connected" - they may be referring to audio interface usage.
Hmm, indeed, it could be read that way. But I guess "The backlight setting is only active during battery operation" doesn't leave much room for different interpretation... I hope you're right, though.
-
[/quote]
I am not sure that necessary is the correct conclusion from what the manual is saying. "Operation on the USB bus" is not the same as "when USB power is connected" - they may be referring to audio interface usage.
[/quote]
I agree, using the deck as a USB interface is different than running a battery via USB, which will show the same screen and same functionality as using an internal battery. As a USB interface, the function of the machine is very different. I do not think the screen will stay lit with an external battery. Just a guess obviously.
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Before discussion is going on with the wrong text, this is what is printed in the manual;
NOTE
The backlight setting is only active during battery operation.
The backlight will always stay lit when operating on USB bus
power.
The last line with "power" was omitted. So its USB bus power.
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What’s the betting that in reality it’s nearer to the Mixpre 3 battery life of 2.5 hours?
Power usage MixPre 3 is with phantom power switched on about 6Watt (not correct, its much higher). Maximum power usage F3 is 5watt. Maximum power usage Tascam FR-AV2 is 3.8Watt. The reason that the power usage of the Zoom F3 is higher has to do that it only has 2 AA batteries for power, ie maximum 3Volt. Tascam has 3AA for power, ie maximum 4.5Volt. To give you an idea what's happening, lets take the 48Volt for phantom power. The microphone uses 3mA with 48Volt. That is 0.144Watt. For the Zoom F3 this will be at least 0.144Watt / 3Volt = 48mA for a mic. For the Tascam FR-AV2 this will be at least 0.144Watt / 4.5Volt = 10,7mA. If you use two microphones this means for the Zoom 96mA and for the Tascam 21.4mA, difference 74.6mA.
Just want to point out a math error in the current flow calculation on phantom channels for the FR-AV2. It works out to 32ma, not 10.7ma, so the difference between the Zoom and Tascam for a 3ma stereo pair is 32ma, not 74.6ma.
Another comment is on power consumption, which is a complex subject as it concerns DC/DC conversion efficiency, etc. But as we know, wattage is a measure of power consumption - that is current times voltage. Higher battery voltage, lower current draw. So, wattage/power consumption only indirectly relates to the number of batteries or their total voltage. The obvious difference between having two cells versus three cells is actual stored energy capacity, which will be approximately 50% larger in the Tascam. All things would indicate the Tascam will have longer running times with internal battery due to the extra cell, and its lower power consumption. In any case, I personally don’t trust my recorders (with the exception of my never-die Sony PCM-M10) to run for long periods on the internal battery. I always use power banks.
-
On paper, this looks really great and may be a better alternative to the Zoom F3.
I am most interested to see how its autoranging ADCs perform, and if they suffer from the same windowing noise modulation as the Zoom F-series does.
-
I am most interested to see how its autoranging ADCs perform, and if they suffer from the same windowing noise modulation as the Zoom F-series does.
How does it perform in the Portacapture X6 and X8?
I’d imagine we should expect something similar.
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On paper, this looks really great and may be a better alternative to the Zoom F3.
I am most interested to see how its autoranging ADCs perform, and if they suffer from the same windowing noise modulation as the Zoom F-series does.
A noise that no one can hear, I believe. At least, I cannot.
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On paper, this looks really great and may be a better alternative to the Zoom F3.
I am most interested to see how its autoranging ADCs perform, and if they suffer from the same windowing noise modulation as the Zoom F-series does.
A noise that no one can hear, I believe. At least, I cannot.
Seconded. And I'm sure most others will agree with. This new device might be worthwhile for somebody who doesn't own an F3 but I can't see myself jumping to buy one to replace what I already own. Just used my F3 for a stealth job last weekend and it serves me perfectly fine.
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On paper, this looks really great and may be a better alternative to the Zoom F3.
I am most interested to see how its autoranging ADCs perform, and if they suffer from the same windowing noise modulation as the Zoom F-series does.
A noise that no one can hear, I believe. At least, I cannot.
Seconded. And I'm sure most others will agree with. This new device might be worthwhile for somebody who doesn't own an F3 but I can't see myself jumping to buy one to replace what I already own. Just used my F3 for a stealth job last weekend and it serves me perfectly fine.
Concur
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Before discussion is going on with the wrong text, this is what is printed in the manual;
NOTE
The backlight setting is only active during battery operation.
The backlight will always stay lit when operating on USB bus power.
Thank you Joop for clearing up the confusion. So it seems the backlight will always light up when a USB battery is attached, regardless of the user preference for backlight. Still hope Tascam reads this and will update the firmware to honor the user backlight setting regardless of the used power source.
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Just want to point out a math error in the current flow calculation on phantom channels for the FR-AV2. It works out to 32ma, not 10.7ma, so the difference between the Zoom and Tascam for a 3ma stereo pair is 32ma, not 74.6ma.
Huh? My bad, you are right, don't know what's happened in my calculator, I can't get the 10.7mA again.
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They're in stock at BHP. Picked up 2. Will run stage/SBD next weekend. Hoping that Bluetooth 5 dedicated dongle (vs 4 on the A10) will prove to be a bit more reliable, I sometimes have issues checking SBD levels via Bluetooth from the front of the room. I'm a little bit skeeved out by not having internal memory to fall back on, too, but I know it'll be fine. Will report back on ease of use.
Edit: I'm just going crazy, I also picked up an Ultrasync Blue. Should be the ultimate quality of life upgrade. Timecode sync, no worrying about levels, more stable Bluetooth monitoring (which has saved my ass in the event I don't see levels at all once the show starts). Ahh. Fingers crossed.
-
Got an email from B&H saying expected home delivery is Oct. 2.
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Got one from B&H today. Due to Rosh Hashana closure the delivery date is October 9th which is no big deal I don't plan to do any shows until after that anyway.
Hope to be able to use this instead of my Oade modded DR100mkii which is just a little too big sometimes and has a fussy internal battery scheme.
After a couple of trial runs I'll try to use it for Bob Weir and the Louisiana Philharmonic in November at the Saenger Theater in NOLA. This plus my MBHO actives will be a jacket pocket sized rig that could be run from seat or clamped and left to run unattended.
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Curious if anyone can lend some insight on the following:
PCM A-10 AUDIO in specifications:
Input impedance: approx. 4.7 kΩ
Rated input level: approx. 2 V
Minimum input level: approx. 18 mV
Tascam FR-AV2 max input specifications:
XLR/TRS LINE level:
24 dBu
3.5MM:
+1 dBV
In trying to shortcut understanding these values, I plugged them into Claude and asked for some details translating them. I came out with the following:
+1 dBV (FR-AV2 3.5mm jack) = 1.122 volts RMS
+24 dBu(FR-AV2 XLR/TRS jack) = 12.28 volts RMS
+8.2 dBu(PCM-A10 LINE IN) = 2.000 volts RMS
Does this seem right? I'm trying to understand the practical applications as it relates to an increase or decrease in headroom, using some DPA 4061s into the FR-AV2 3.5mm as opposed to the PCM-A10. At this point all I know is "well, one of those numbers is bigger than the other!", but in terms of practical applications, with the goal being "I don't want to have to worry as much about digital peaking in my recording device", I'm trying to gain further understanding.
Seems that I may be best served using something like this?
https://immersivesoundscapes.com/accessories/Adapter-for-PIP-mics-to-work-on-XLR-48V-Recorder-p492004466
Do the benefits afforded by dual ADC outweigh any concern related to max input with those values?
These are the questions I'm asking myself and attempting to research and understand. Just wanted to pose the question here in case anyone is able to quickly lend some insight. These values and what they mean are a bit of new territory for me.
edit: Here's the conclusion I'm running with for the time being. Will ultimately test both in the field side by side. If anyone wants to chime in with why this is bullshit or correct, feel free. Just output from discussions with LLM (Claude)
To answer this question, we need to compare the maximum input levels and the recording technology of both devices:
PCM-A10 LINE IN: +8.2 dBu (2.000 volts RMS)
FR-AV2 3.5mm jack: +1 dBV (1.122 volts RMS)
At first glance, the PCM-A10 appears to be the "safer" choice because it can handle a higher input level before clipping in its analog stage. However, this doesn't tell the whole story. Let's break it down further:
FR-AV2 advantages:
32-bit float recording: This provides an enormous dynamic range, theoretically capturing everything from the noise floor to well above 0 dBFS.
Dual ADC system: One ADC captures the full range, while another captures at a lower gain, providing a safety net for unexpected peaks.
PCM-A10 limitations:
Traditional fixed-point recording (likely 16 or 24-bit).
Single ADC system.
Why the FR-AV2 might be safer despite the lower maximum input level:
Even if the analog input stage clips, the lower-gain ADC in the dual system can still capture the peaks accurately.
The 32-bit float format allows for extreme gain adjustments in post-production without introducing quantization noise.
You have more flexibility in setting levels. You can set a lower input gain to avoid clipping the analog stage, and still capture quiet details thanks to the high bit depth.
In practical terms:
With the PCM-A10, if you clip the input, that information is lost forever.
With the FR-AV2, even if you clip the analog input stage, you're likely to still have usable audio data that can be recovered in post-production.
Conclusion:
The FR-AV2 with its 3.5mm input would likely be the safer choice, despite its lower maximum input level on paper. The advanced recording technology provides a much wider margin for error and more forgiving post-production options.
That said, the ideal solution would be to use the XLR/TRS inputs on the FR-AV2 if possible, as they provide even more headroom. If you must use the 3.5mm input, start with a lower gain setting than you would on the PCM-A10, and trust in the 32-bit float and dual ADC system to capture the full dynamic range of your source.
Remember to always do test recordings in conditions similar to your actual recording scenario to ensure you're getting the results you need.
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Curious if anyone can lend some insight on the following:
PCM A-10 AUDIO in specifications:
Input impedance: approx. 4.7 kΩ
Rated input level: approx. 2 V
Minimum input level: approx. 18 mV
The problem with LLMs is that they are good at summarizing text. But they can't think. And they can't make sense of what they output in the same way any human child can. It simply repeats what it has been trained with. In this case: marketing materials as it seems. ;)
You write that you're running DPA 4061s. These have a sensitivity of 6 mV per Pa (that's Pascals, the unit for measuring atmospheric pressure). The max. input level of 2 V on the A10 gives you 2000 mV / (6 mV / Pa) = 333.3 Pa. That equals approx. 144 dB SPL (here's a calculator (https://sengpielaudio.com/calculator-soundlevel.htm)). This means the input stage of your A10 will clip when your DPAs are recording sounds louder than 144 dB. These are SPLs you will find at rocket launches, but not when recording music. So you're safe with the A10.
You can repeat the same calculations with the other values. 1 dbV = 1.12 V and 24 dbU = 12.27 V (calculators (https://sengpielaudio.com/calculator-db-volt.htm)).
1.12 V / (6 mV / Pa) = 186 Pa (= 139 db SPL)
12.27 V / (6 mV / Pa) = 2045 Pa (= 160 dB SPL)
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Curious if anyone can lend some insight on the following:
PCM A-10 AUDIO in specifications:
Input impedance: approx. 4.7 kΩ
Rated input level: approx. 2 V
Minimum input level: approx. 18 mV
The problem with LLMs is that they are good at summarizing text. But they can't think. And they can't make sense of what they output in the same way any human child can. It simply repeats what it has been trained with. In this case: marketing materials as it seems. ;)
You write that you're running DPA 4061s. These have a sensitivity of 6 mV per Pa (that's Pascals, the unit for measuring atmospheric pressure). The max. input level of 2 V on the A10 gives you 2000 mV / (6 mV / Pa) = 333.3 Pa. That equals approx. 144 dB SPL (here's a calculator (https://sengpielaudio.com/calculator-soundlevel.htm)). This means the input stage of your A10 will clip when your DPAs are recording sounds louder than 144 dB. These are SPLs you will find at rocket launches, but not when recording music. So you're safe with the A10.
You can repeat the same calculations with the other values. 1 dbV = 1.12 V and 24 dbU = 12.27 V (calculators (https://sengpielaudio.com/calculator-db-volt.htm)).
1.12 V / (6 mV / Pa) = 186 Pa (= 139 db SPL)
12.27 V / (6 mV / Pa) = 2045 Pa (= 160 dB SPL)
Appreciate those calculators and details, thanks.
Did you mean to say "you are safer" with the A10? I know I'd be fine, it has been my primary device for awhile, what I'm trying to figure out is the actual practical benefit I'd receive from 32bit float / ADC of the FR-AV2 given the lower max input level of the 3.5mm jack.
In the 24 bit/A10 "world", let's call it, I am using a preamp to apply gain and get as loud as possible without clipping so that I don't have a hissy recording (if I don't send the signal to the recorder with enough gain, and raise in post, I'll likely get hiss)
In the FR-AV2 world, I don't have to be concerned with digital clipping on the device -- I can essentially run as hot as I want and lower after the fact or so I thought. This "max input level" is throwing me for a bit of a loop. In a practical sense, does it mean that if I apply too much gain via the preamp, I am screwed? No amount of post processing could repair the exceeded input levels, right?
But the thing is, I shouldn't really have to worry about applying gain much at all.. I can apply little to no gain, set conservative levels and thanks to the ADC almost certainly not have to worry about hiss if I have to bring it up a bunch later. This is where I am landing in theory -- will see the reality once I am able to actually test. I am suspecting that in effect it means I do still need to worry about setting levels for louder/quieter shows, depending upon how noisy the pres are. Unless I use the XLR/TRS. There, I actually have enough headroom that I can more or less set it up once and never worry about it again.
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In the FR-AV2 world, I don't have to be concerned with digital clipping on the device -- I can essentially run as hot as I want and lower after the fact or so I thought. This "max input level" is throwing me for a bit of a loop. In a practical sense, does it mean that if I apply too much gain via the preamp, I am screwed? No amount of post processing could repair the exceeded input levels, right?
You're right you don't have to be concerned with digital clipping, but that doesn't mean you can run as hot as you want. If you run it too hot, you will overload the analogue stage of the recorder - you still won't get digital clipping, but you very much will get analogue distortion.
The idea behind the better 32-bit float recorders (the MixPre II, Zoom F3/F6, Tascam FR-AV2, etc, as opposed to for example the Zoom H1essential) is that you won't have to use an external preamp. They all have very clean internal ones.
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Cool, thanks. I knew this was a concern, I just wasn't expecting the max input to actually be lower than the A10. Was surprised once I did the conversions and realized. Fingers crossed they're as clean as we hope.
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Have you actually tried running lower and raising the levels to see if you get hiss? With DAT era devices, that was definitely needed, but I regularly record at -30db and then normalize, and haven't had a problem of noisy recordings. The main difference with 32 bit devices is the recorder basically forces you to run low.
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In the FR-AV2 world, I don't have to be concerned with digital clipping on the device -- I can essentially run as hot as I want and lower after the fact or so I thought. This "max input level" is throwing me for a bit of a loop. In a practical sense, does it mean that if I apply too much gain via the preamp, I am screwed? No amount of post processing could repair the exceeded input levels, right?
You're right you don't have to be concerned with digital clipping, but that doesn't mean you can run as hot as you want. If you run it too hot, you will overload the analogue stage of the recorder - you still won't get digital clipping, but you very much will get analogue distortion.
The idea behind the better 32-bit float recorders (the MixPre II, Zoom F3/F6, Tascam FR-AV2, etc, as opposed to for example the Zoom H1essential) is that you won't have to use an external preamp. They all have very clean internal ones.
Do we have good info on the quality of the Tascam FR-AV2's pre-amps?
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I'd be particularly interested in recording level functionality when using 32 bit float. With the F3 there is a fixed gain in the analog input, and then the 32 bit float samples are stored with an amplification depending on the 'magnification level' as was set at start of recording.
I can't find any details in the FR-AV2 manual about something like this. It seems as if the record level still applies to the analog input, regardless if 24bit or 32bit float is used. This would make it still possible to set the record level 'too high' when using 32bit float, resulting in overloading the analog input stage, kind of defeating the merrits of a 32 bits float recorder (set-and-forget). I might be wrong, of course...
Hope to hear soon about real-life experiences with this interesting device! I'm about to replace my Edirol R-09HR, and it seems I will choose between the F3 and the FR-AV2... :headphones:
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Do we have good info on the quality of the Tascam FR-AV2's pre-amps?
The info available so far says that it has Tascam's HDDA mic preamps that are presumably the same discrete design as the 701d and some other devices in their family. Here's the marketing speak from the 701d page -
"HDDA mic preamps - the differential mic preamp circuits use discrete architecture, built with carefully selected components. For example, Texas Instruments OPA1652 op amps were selected for their high quality and low noise – with even better audio performance than the previous DR-70D model. As a result, the unit boasts an equivalent input noise of -124 dBu or better"
I ran a 701d for a few years and it sounds pretty good. The preamps are just fine for what many users are going to do - live music, film dialog, etc.
There's a bunch of recordings in my LMA signature link that use the 701d preamps if you want to have a listen in different scenarios (inside, onstage, outside, matrix) and with different mics (I had several preamps at the time and often used them in front of the recorder but there's some with the stock device Pres in there). That device is marketed towards videographers like this new one is and has many of the same features.
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Have you actually tried running lower and raising the levels to see if you get hiss? With DAT era devices, that was definitely needed, but I regularly record at -30db and then normalize, and haven't had a problem of noisy recordings. The main difference with 32 bit devices is the recorder basically forces you to run low.
Yes e.g. https://ia802308.us.archive.org/4/items/tmg2021-09-23/05_Hair%20Match.mp3
I think this may have been running my DR680MKii low as opposed to A10, but same principle. Certainly didn't kill the tape or anything, just something I try to avoid since.
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I would very much hesitate to run at -30dB even at 24 bit. The noise floor will come up with the music when you amplify it in post. I tend to shoot for -14 to -12, even -16 if I think the band is super loud and may amplify by 6 dB at some point, especially at the end of the show.
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SBD > dr680mkii + CA-14 omni (healy on stage) > CA-Ugly 2 > pcm m10
So the noise could have come from the soundboard, the ca-ugly, the m10, something done in the mix, etc.
The 680mk2 has 100db SNR, so amplifying the noise floor by 30db should still have the noise floor of the recorder at -70db. Once you add an external preamp into the mix, you need to determine the right levels for the preamp and recorder to minimize noise, as that's a lot different for every device.
Given that most 32bit devices are applying 0 analog gain, if you're recording music that doesn't incorporate a jet that's taking off, you're running at significantly lower than -30db peaks.
Have you actually tried running lower and raising the levels to see if you get hiss? With DAT era devices, that was definitely needed, but I regularly record at -30db and then normalize, and haven't had a problem of noisy recordings. The main difference with 32 bit devices is the recorder basically forces you to run low.
Yes e.g. https://ia802308.us.archive.org/4/items/tmg2021-09-23/05_Hair%20Match.mp3
I think this may have been running my DR680MKii low as opposed to A10, but same principle. Certainly didn't kill the tape or anything, just something I try to avoid since.
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Given that most 32bit devices are applying 0 analog gain, if you're recording music that doesn't incorporate a jet that's taking off, you're running at significantly lower than -30db peaks.
I'm not sure this is true? Just because you don't set the gain, that doesn't mean 0 analogue gain is applied. Take this from the Zoom F3 specs:
Equivalent input noise: −127 dBu or less (IHF-A) when waveform magnification is ×1024 with 150 Ω input.
As far as I can tell, this sort of EIN can usually only be attained when your gain is of at least +30 dB. If you take a look at this table (https://avisoft.com/recorder-tests/), you will notice the EIN is always measured at the maximum available gain settings, which in the F3 is equivalent to the waveform magnification of ×1024.
A recorder with no gain will never have an EIN of -127 dBu. The converter's self-noise normally sits around -95 dBu. What improves the EIN drastically is the use of a preamp, which does add a little noise, but proportionally adds a LOT more gain than noise. So if the preamp adds 30 dB of gain and only 5 dB of noise, then your EIN will be -95 +5 -30 = -120 dBU
AFAIK, the trick with these 32-bit float devices is that they essentially add a lot of clear analogue gain when the signal they receive is weak, and little or no gain when the signal is strong (and then, digitally, they account for the different amounts of gain when merging these two paths).
What I'm trying to say is that using one of these devices is not at all like running -30 dB peaks with a recorder with a single ADC. If you're running -30 dB peaks with a regular recorder, the converter stage self-noise will instantly drop to -65dB or so when you normalise. The peaks themselves likely will be fine, but the quieter parts will suffer for sure.
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I'd be particularly interested in recording level functionality when using 32 bit float. With the F3 there is a fixed gain in the analog input, and then the 32 bit float samples are stored with an amplification depending on the 'magnification level' as was set at start of recording.
I can't find any details in the FR-AV2 manual about something like this. It seems as if the record level still applies to the analog input, regardless if 24bit or 32bit float is used. This would make it still possible to set the record level 'too high' when using 32bit float, resulting in overloading the analog input stage, kind of defeating the merrits of a 32 bits float recorder (set-and-forget). I might be wrong, of course...
Hope to hear soon about real-life experiences with this interesting device! I'm about to replace my Edirol R-09HR, and it seems I will choose between the F3 and the FR-AV2... :headphones:
Just received mine. Just to answer this, with 32 bit float enabled, you can still set gain independently to each channel (1 or 2). From 0 to +60 DB. It is set to +18 by default. Can adjust while recording though the menu is a bit clunky. Expect it's easier with app.
Going to be testing some items for my purposes (battery time recording with my mics, effective Bluetooth range, app controls, running hot with loud stereo approximate concert volume and reducing in post, etc) but if anyone wants any info I may be able to provide feel free.
As an aside, I'm a dumbass and just clicked the suggested cards when purchasing it from BHP, didn't realize it suggested the wrong size. I bought two bigger ones. Need to return em.
Edit: note you need to assign both channels to EXT input manually if you plan to use the 3.5mm. Flip the STEREO LINK toggle under INPUT to do so. No auto detection or stereo assumption - threw me for a loop for a second.
Edit 2: another note, two drawbacks to the lack of explicit HOLD.
1- you hold down 1 button for several seconds to stop the recording. Could conceive of this occurring in a pocket, though unlikely.b
2 - there is one unlocked button that could fuck up the recording, but you'd need to hit that button, then a separate button, then a third different button several times, in a particular order. Again, unlikely, but technically possible. All the buttons are pretty hardy and difficult to press accidentally.
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"with 32 bit float enabled, you can still set gain independently to each channel (1 or 2)". Sorry to be slow, but how is this useful? I own a F3 and I can always adjust the relative volume levels in post as needed.
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As far as I can tell, this sort of EIN can usually only be attained when your gain is of at least +30 dB. If you take a look at this table (https://avisoft.com/recorder-tests/), you will notice the EIN is always measured at the maximum available gain settings, which in the F3 is equivalent to the waveform magnification of ×1024.
A recorder with no gain will never have an EIN of -127 dBu. The converter's self-noise normally sits around -95 dBu. What improves the EIN drastically is the use of a preamp, which does add a little noise, but proportionally adds a LOT more again than noise. So if the preamp adds 30 dB of gain and only 5 dB of noise, then your EIN will be -95 +5 -30 = -120 dBU
This is correct. That's why EIN figures are useless if the conditions under which these were achieved are not known. Tascam lists the EIN as "<= -127 dBu", but doesn't mention at which gain and with which source resistance, or whether or not it is a-weighted. But since they probably want to be able to list the best possible EIN, we can safely assume this was measured with the recorder at max. gain (50 dB). This is a pretty big amplification factor and I guess most people will never really need to apply this much gain. So this measurement does not tell us anything about the pre-amp's performance at, e.g. unity gain. The other parameters remain unknown, so we can't really use their EIN number to compare the pre-amps to other devices.
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"with 32 bit float enabled, you can still set gain independently to each channel (1 or 2)". Sorry to be slow, but how is this useful? I own a F3 and I can always adjust the relative volume levels in post as needed.
I do not see why it would be particularly useful. Maybe if you were running two different mics with wildly different sensitivities? I was just reporting anything relevant to setting levels while recording since TheJez expressed curiosity.
On another note, holy hell is the multi device management & app in general just miles ahead of the A10 app. The only thing that's wonky is adjusting any of the levels - it's a slider rather than a button, and it is sort of 'sticky' so if you are sliding the OUTPUT playback from 10 to 30, you will drag it to 30 and then it will snap back to 10 and rapidly climb up, reflecting the device actually receiving your command to raise the volume and showing you how that is happening in realtime. Have seen it snap back and forth a bit, it can be a bit finicky - I prefer the A10 app's approach to just giving you a button, so you can press however many times to raise it the exact amount you're expecting without even looking, really.
I went into a bit of detail there because it is the singular only complaint I have. Everything else is excellent, and I work on a team developing mobile apps in my professional life, so I know how much of a pain in the ass Android + Bluetooth development is to get right.
The app allows you to adjust basically everything on the device remotely, allows you to monitor levels of multiple devices at once (I experienced intense pain at times switching from device to device when I was running multiple A10s - not an issue, here.) start or stop devices simultaneously, easily remotely monitor with Bluetooth headphones as well as remote control, and it has a HOLD button so you can monitor levels without worrying about phantom touches from sweaty concert gear setup fingers. Ironically monitoring levels probably won't be necessary for the most part anymore, but still. I like to have it out and showing levels particularly for the first note of the show, as I've experienced levels suddenly dropping once they switch over from the house music, at which point I can signal the engineer to save it if they are so kind.
On the Bluetooth connectivity front, I just did a shootout between the A10 and the AV2 and sadly the A10 absolutely smokes it - even on "stable". I was able to get at least double the effective range (~60 vs ~30 ft) out of the A10. Was hoping for the opposite, that's a major damper for me. Extremely surprising given the increased power capacity and Bluetooth 5 of the AV2. Doesn't help not worrying about levels if I can't confirm that I'm receiving a signal :/
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in setting mine up today I noticed that when powering the unit OFF and then ON later...
the Phantom Power setting defaults backs to OFF... ALL other Input settings are saved
the Manual does indicate...
The following input settings can be saved and recalled.
- REC LEVEL
- GANG
- DELAY
- LOW CUT
- DYNAMICS
- EQ (MANUAL EQ)
- NOISE GATE
my other Tascam recorders will ask when powering on... "Phantom Power is ON... are you sure?" or something along those lines
"with 32 bit float enabled, you can still set gain independently to each channel (1 or 2)". Sorry to be slow, but how is this useful? I own a F3 and I can always adjust the relative volume levels in post as needed.
in the Input settings from 0 to +60dB and you can Gang the channels or set separately
being able to set levels at the Input stage avoids accidental overloads that CANNOT be fixed
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Oh OK, so the gain setting applies to the input stage before the signal hits the floating 32 bit innards of the device.
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Just received mine. Just to answer this, with 32 bit float enabled, you can still set gain independently to each channel (1 or 2). From 0 to +60 DB. It is set to +18 by default. Can adjust while recording though the menu is a bit clunky. Expect it's easier with app.
Is it ONLY in 32 bit mode where you can independently set levels, or can you also do that in 24 bit as well?
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in setting mine up today I noticed that when powering the unit OFF and then ON later...
the Phantom Power setting defaults backs to OFF... ALL other Input settings are saved
the Manual does indicate...
The following input settings can be saved and recalled.
- REC LEVEL
- GANG
- DELAY
- LOW CUT
- DYNAMICS
- EQ (MANUAL EQ)
- NOISE GATE
my other Tascam recorders will ask when powering on... "Phantom Power is ON... are you sure?" or something along those lines
"with 32 bit float enabled, you can still set gain independently to each channel (1 or 2)". Sorry to be slow, but how is this useful? I own a F3 and I can always adjust the relative volume levels in post as needed.
in the Input settings from 0 to +60dB and you can Gang the channels or set separately
being able to set levels at the Input stage avoids accidental overloads that CANNOT be fixed
This issue seems to be fixed in the new firmware 1.01, see https://tascam.jp/int/product/fr-av2/support (https://tascam.jp/int/product/fr-av2/support)
The phantom power setting would always be turned off when the power was turned on. This has been fixed.
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Well. Bad news.
I left it recording for a few hours, just some CA-14s and DPA 4061s into the 3.5mm both powered by their own battery boxes, PIP off, and for some reason, now, the there is a click in the left channel every ~.5 seconds.
Worse in one device than the other for some reason.
Can anyone else with one of these please perform the following steps:
Record for at least 3 hours with the 3.5mm. Then..
Set input to EXT (ST)
Set REC LEVEL to +60 (this is merely to see the issue more clearly, it is present no matter what)
Do not have ANYTHING plugged in.
Observe.
This is what I am getting, on both of my devices. Here is an example of the raw audio, as well as a video of what it looks like, and showing that I have absolutely nothing whatsoever plugged in:
https://drive.google.com/file/d/1TZzY6AXcXIu4Se0-kTGoGQqxjCsRuQt5/view?usp=sharing
https://drive.google.com/file/d/1MikyxW8Oo2ziBg-sY6JnhR4mWH-QhqiQ/view?usp=sharing
This issue is present no matter what the input source is, now. Even if it's set to MIC, it's still there, albeit much lower, given the much higher headroom of the XLR/TRS inputs.
I will see if factory resetting or firmware update does anything, but.... this seems like the devices are just defective, right?
Edit: Updated firmware, issue still present.
Edit 2: Factory reset, issue still present. I originally thought it went away because of the reset levels. Nope, still there in full force.
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Move the battery boxes and mics away from the recorder and see if the issue disappears. If it does, then it's RFI from the recorder (most likely Bluetooth). Or try turning every thing off in the recorder that could remotely emit an electromagnetic field.
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Move the battery boxes and mics away from the recorder and see if the issue disappears. If it does, then it's RFI from the recorder (most likely Bluetooth). Or try turning every thing off in the recorder that could remotely emit an electromagnetic field.
Walking around my home (Just holding the AV2, nothing at all plugged in or near) it is perfectly consistent. Bluetooth is off after factory reset.
Here is an example actually recording audio:
https://drive.google.com/file/d/1dsvI7OdnX9d_mR2Uax-CkZEfMJ2rLb1_/view?usp=drive_link
This is me recording with the default levels (+18), battery box (no external gain) and talking about 2 feet away from the mic. Then normalizing it.
For our purposes, this would probably be moot. Would likely be completely inaudible with the relative noise floor which would be present at vast majority of live acts. An external preamp would negate the concern.
Using XLR/TRS would also just about guarantee it'd never be an issue.
Doesn't concern me quite so much after doing this as a test but... I don't know. When I was first testing, I turned the gain up pretty damn high and just listened to ambient sound, and I did not hear the clicks. Perhaps the ambient volume of my living room was a fair bit higher than the bedroom I'm in now, fan on higher setting, A/C on... I am suspecting it's always there, and it will not actually be an issue recording typical volumes.
If anyone else can just set both channels to EXT and crank it up to +60 and let me know if you see the same thing, I'd appreciate it. Now I'm suspecting it's just something internal to the device.
I was planning to run 4061s into the AV2 with just a BB and do a shootout with my typical CA-UGLY > CA-14 > A10 chain, but now I'm thinking I'll want to run DPA 4060s instead. To ensure there is no chance that I ever have to worry about those clicks..
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I'd be particularly interested in recording level functionality when using 32 bit float. With the F3 there is a fixed gain in the analog input, and then the 32 bit float samples are stored with an amplification depending on the 'magnification level' as was set at start of recording.
I can't find any details in the FR-AV2 manual about something like this. It seems as if the record level still applies to the analog input, regardless if 24bit or 32bit float is used. This would make it still possible to set the record level 'too high' when using 32bit float, resulting in overloading the analog input stage, kind of defeating the merrits of a 32 bits float recorder (set-and-forget). I might be wrong, of course...
Hope to hear soon about real-life experiences with this interesting device! I'm about to replace my Edirol R-09HR, and it seems I will choose between the F3 and the FR-AV2... :headphones:
Just received mine. Just to answer this, with 32 bit float enabled, you can still set gain independently to each channel (1 or 2). From 0 to +60 DB. It is set to +18 by default. Can adjust while recording though the menu is a bit clunky. Expect it's easier with app.
Thanks for testing this. I guess it's still not fully clear if in 32bit float the gain is applied in the analog stage or in the digital domain. I guess it's in the analog stage, which would be a major difference compared to the F3. If so, this one seems to leave it to the user to set the gain 'good enough', 'optimal' or 'completely wrong'... :-\
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I'd be particularly interested in recording level functionality when using 32 bit float. With the F3 there is a fixed gain in the analog input, and then the 32 bit float samples are stored with an amplification depending on the 'magnification level' as was set at start of recording.
I can't find any details in the FR-AV2 manual about something like this. It seems as if the record level still applies to the analog input, regardless if 24bit or 32bit float is used. This would make it still possible to set the record level 'too high' when using 32bit float, resulting in overloading the analog input stage, kind of defeating the merrits of a 32 bits float recorder (set-and-forget). I might be wrong, of course...
Hope to hear soon about real-life experiences with this interesting device! I'm about to replace my Edirol R-09HR, and it seems I will choose between the F3 and the FR-AV2... :headphones:
Just received mine. Just to answer this, with 32 bit float enabled, you can still set gain independently to each channel (1 or 2). From 0 to +60 DB. It is set to +18 by default. Can adjust while recording though the menu is a bit clunky. Expect it's easier with app.
Thanks for testing this. I guess it's still not fully clear if in 32bit float the gain is applied in the analog stage or in the digital domain. I guess it's in the analog stage, which would be a major difference compared to the F3. If so, this one seems to leave it to the user to set the gain 'good enough', 'optimal' or 'completely wrong'... :-\
Not an expert or close to it, but it would make no sense to me to have gain adjustments applied to a floating 32 bit-equipped recorder in the digital domain. A previous response to my earlier dumb question suggested to me that the gain adjustments for this device applied to the analogue stage upstream from the digital domain. FWIW, I have only encountered overloading the pre-digital domain (i.e., the analogue input stage) with my F3s when very strong gusts of air were blown directly into my unshielded (dumb mistake on my part) Line Audio CM3 mics--once by massive stage subwoofers, once by being way too close (a foot) to a bass drum, and then again at home when I blew hard into them. Wiser positioning and windscreens have addressed that issue for me.
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Move the battery boxes and mics away from the recorder and see if the issue disappears. If it does, then it's RFI from the recorder (most likely Bluetooth). Or try turning every thing off in the recorder that could remotely emit an electromagnetic field.
I have avoided using Bluetooth with my devices for fear of RFI, although more experienced folks here have reassured me that they have not encountered RFI with devices connected to their cell phones via Bluetooth. However, this post suggests that RFI in this case may be an issue? FWIW, I often use an F3 equipped with a XLR-minijack Y adapter with a nine volt SP battery box and CA-14 mics (with no power to the mics from the F3) and have never had an issue with clicks or RFI.
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It's not RFI, I am reasonably confident that anyone would see/hear it if they set input to EXT and applied max gain, or boosted a recording enough in a DAW.
I am pretty confident now that it's basically a non issue for 99% of applications. Only if you had an extremely low sensitivity mic, and you were recording a relatively quiet source, through 3.5mm, would you need to apply so much gain that it would be in any way audible.
Seems to just be internal mechanisms exposed in the edge case of applying extreme gain to the noisiest setting (EXT input).
I was more worried when it seemed as though it only appeared after a bit of use -- I now think it was always there and that I just didn't notice it in the room I was initially testing extreme gain / ambient noise in. If someone else can confirm that they see identical behavior when just setting to EXT +60 with nothing plugged in, as in the video above, it will all but confirm it.
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Set REC LEVEL to +60 (this is merely to see the issue more clearly, it is present no matter what)
Do not have ANYTHING plugged in.
So you have an open input which is set to maximum amplification. It will pickup hum and alike. That's what you see. You have the same problems when the input is properly loaded? When you have a ticking sound with regular intervals you probably have a problem with static discharge. It can come from anything. Placed the recorder on a rubber isolation to rule out your table or you? May be there is a high voltage power line in your direct area. I don't think its a malfunction of the recorder as you have two examples, but it could be that something did go wrong at the assembly. In any case if this is persistent I would go back to your supplier.
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Set REC LEVEL to +60 (this is merely to see the issue more clearly, it is present no matter what)
Do not have ANYTHING plugged in.
So you have an open input which is set to maximum amplification. It will pickup hum and alike. That's what you see. You have the same problems when the input is properly loaded? When you have a ticking sound with regular intervals you probably have a problem with static discharge. It can come from anything. Placed the recorder on a rubber isolation to rule out your table or you? May be there is a high voltage power line in your direct area. I don't think its a malfunction of the recorder as you have two examples, but it could be that something did go wrong at the assembly. In any case if this is persistent I would go back to your supplier.
No. Please see attached video above. The only thing being recorded there is the device itself, which does not have onboard microphones. No mics. No pip. No phantom power. Just 3 AA batteries. And the device.
Yes I've ruled out external factors.
I would return it if it wasn't the expected behavior of all devices, which is what I suspect. I've opened a ticket with Tascam to confirm. Others here with AV2s should be able to confirm that they see the same which will just about seal it, though.
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Well. Bad news.
Just a thought.
Did you power it with batteries, USB-C or with an external power supply?
The sound reminds me of voltage/power regulation.
What if you try usb-c from different sources or a couple of different battery chemistries?
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Well. Bad news.
Just a thought.
Did you power it with batteries, USB-C or with an external power supply?
The sound reminds me of voltage/power regulation.
What if you try usb-c from different sources or a couple of different battery chemistries?
Yep good thought. I did remove the batteries and confirm same behavior with USBC power source. Given the identical result I ruled out the power source being the root.
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This is without any mics connected? Good thing I won't be using it to record with nothing connected to it.
I may have missed something but did these mystery clicks happen with mics attached to the device the same way it did with nothing connected?
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This is without any mics connected? Good thing I won't be using it to record with nothing connected to it.
I may have missed something but did these mystery clicks happen with mics attached to the device the same way it did with nothing connected?
Yes. I perceive snark, I don't really understand why. I was just trying to clearly communicate a potential issue with the device, and it's important to know if it's coming from the device, or in any way related to mics being attached. The fact that it is audible without anything else attached, as well as with mics attached, is important information.
I already posted an example of the raw audio of the issue alone, video showcasing the issue, described the reproduction steps and settings utilized, and an example of it audible while recording with mics.
Through this process determined that it should not be an issue for our purposes, as I stated. Still a potential issue if someone wants to record quiet sources with the 3.5mm input without a preamp.
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One test I would try if I had one of the machines is recording with plug-in-power turned OFF. Mics plugged in would present close to the nominal expected input source impedance, even if they aren’t powered. Clean plug-in-power voltage is even more important on an unbalanced input than pure phantom voltage is on a balanced input. The other test I might try is with the input(s) shorted (also with P-i-P off) to see if the noise still presents. Periodic clicks or ticks are usually clock related leakage from the digital side to the analog side of the machine. Tascam incorporates shielding between these boards on even their most inexpensive recorder, so it could be a matter of a simple production fix if there’s a leak somewhere. This can hopefully be applied to existing machines if it’s deemed to be serious enough.
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Kyle already stated that he tested it with an external battery box + mics and PIP off. (see this post (https://taperssection.com/index.php?topic=205834.msg2418595#msg2418595))
Another thing that should be tested are different bit depths/sampling rates. Could be a timing issue with the ADC.
Also, Kyle, did you happen to test it with a line source (e.g. anything that's not a microphone)? Or even better: Just a single 50 - 300 Ω resistor across one channel and ground?
I have avoided using Bluetooth with my devices for fear of RFI, although more experienced folks here have reassured me that they have not encountered RFI with devices connected to their cell phones via Bluetooth. However, this post suggests that RFI in this case may be an issue? FWIW, I often use an F3 equipped with a XLR-minijack Y adapter with a nine volt SP battery box and CA-14 mics (with no power to the mics from the F3) and have never had an issue with clicks or RFI.
I neither have an F3 nor this Tascam recorder, but my Sony PCM-A10 is emitting some sort of RFI when its NFC feature is enabled and it is placed directly near a not-so-well shielded cable and when gain is set really high. This is only NFC, though. Never had the same problem with pure Bluetooth (with Bluetooth on and NFC off). I guess NFC has a stronger signal or simply broadcasts at a different frequency. That's why I thought it would be wise to check if this is a problem with the Tascam unit. But given what Kyle reports, it doesn't seem to be a problem in his case as he has the same clicks with Bluetooth off.
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SNIP < Yes. I perceive snark, I don't really understand why. I was just trying to clearly communicate a potential issue with the device, and it's important to know if it's coming from the device, or in any way related to mics being attached. The fact that it is audible without anything else attached, as well as with mics attached, is important information.> END SNIP
Relax, bro. This thread got really long really fast with several people chiming in I just asked for clarification since I have one of these on the way.
I didn't, nor do I have plans to, go through all the linked samples and videos and whatnot especially if they are recordings of nothing attached to record with. That's why I asked to clarify if this hiccup is with mics attached, without or both. If it's without I can disregard since it doesn't matter to me if the device has some kind of clicky sounds recording nothing - since I have no plans to record nothing.
Thanks for your early testing and reporting on this device.
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Kyle already stated that he tested it with an external battery box + mics and PIP off. (see this post (https://taperssection.com/index.php?topic=205834.msg2418595#msg2418595))
Another thing that should be tested are different bit depths/sampling rates. Could be a timing issue with the ADC.
Also, Kyle, did you happen to test it with a line source (e.g. anything that's not a microphone)? Or even better: Just a single 50 - 300 Ω resistor across one channel and ground?
Hi Sebastian. Yes I saw that post, but believe that is different than what I am suggesting since the external battery box is an unnecessary variable. Totally concur with your suggestions, though!
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Periodic clicks or ticks are usually clock related leakage from the digital side to the analog side of the machine. Tascam incorporates shielding between these boards on even their most inexpensive recorder, so it could be a matter of a simple production fix if there’s a leak somewhere. This can hopefully be applied to existing machines if it’s deemed to be serious enough.
It is 100% this.
I can hear it shifting as I adjust the "DELAY" setting.
Tested all the ideas posted (24 bit, 96/192khz, PIP off with 3.5mm MIC plugged in), still present in all cases, only change was that the levels showed a bit higher with each tick @ 192 khz.
Curious how TASCAM will respond, I will attempt to clarify with them to see if this is expected from their end. So far they told me to update the firmware, and when I replied that I already did, they sent a link to update the firmware haha.
Again, just to clarify, I am no longer worried about this for real world applications -- I was far more concerned when I thought it started up out of nowhere, since it seemed like there was risk of it worsening.
At these levels, should never hear the internal clock on an actual tape.
Oh and @ rastasean, sorry for late response, yes, can independently adjust channel levels at all formats / bit depths.
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Periodic clicks or ticks are usually clock related leakage from the digital side to the analog side of the machine…
<humor>
Tascam do call the device for a Recoder in the metadata - so you might be onto something :)
</humor>
Producent: TASCAM PCM Recoder FR-AV2
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Just realized another major negative for me. Unless I am missing something, there is no way to turn off the screen. You can turn off the "backlight", but even with that off, the screen still emits light. You can flip it over but then you run the risk accidentally depressing the button that stops recording.
Edit: wait, what the hell, the screen turns off after 30 seconds while recording in POWER SAVE mode. But you cannot turn on phantom power in this mode. I can't figure out a way to get screen off while recording + phantom power. Surely that can't be right!? Imagine some sort of fix will come in a firmware update unless I'm missing it.
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Do sound devices mxpre II recorders completely turn off the display during recording?
The disadvantage of color displays may just be a fact of life that they're 'always on'; I don't know. The most recent recorder I have is the Sony A10 and it doesn't have a color display , but is 'always on' during recording; no determinable light is emitted from it, though.
Another option, instead of turning it over, is to cover the display. Not necessarily practical, but maybe more realistic than turn it over on its face, where I would be concerned about scratching the display.
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You can turn off the "backlight”
Well, only when powering with batteries. When powering through usb-c, you can’t switch off the backlight :(
This display attracting attention is a concern with me as well…
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Yeah, covering screen with some gaffer's tape (or whatever) and controlling via the app is the way to go if you prefer discretion. Just so bizarre that there's a mode which CAN turn the screen off entirely as you'd like, but ONLY without running phantom power.
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Yeah, covering screen with some gaffer's tape (or whatever) and controlling via the app is the way to go if you prefer discretion. Just so bizarre that there's a mode which CAN turn the screen off entirely as you'd like, but ONLY without running phantom power.
Fortunately these kind of weird ‘features’ seem like things that could be fixed by firmware updates. Not sure how Tascam responds to user feedback…
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It seems like the brave first movers are getting their decks and discovering quirks, faults or inconveniences that may or may not be resolved in future firmware updates.
We know for certain that many, if not most, products are released prematurely and fixed or improved with firmware updates later.
Therefore: A friendly suggestion to those who post their fault observations about their new deck:
-It would be very helpful if you include firmware version number when reporting an issue.
- If you have bluetooth connectivity issues, a smartphone brand and OS details could also be useful information.
- If a specific issue give reason to suspect hardware issues, a serial number could also be relevant for future attempt to isolate a range of units with errors.
When including the FW number it may help to identify if a poster is struggling with issues that a more current FW version has resolved.
It will also help future readers of the early posts in this thread; it will be easier identify if a discussed problem may have been resolved through FW updates at the time in the future when reading the post.
These device threads often evolve to become so long that we don't read them end-to-end but access them selectively through search methods.
The FW version number would a convenient indicator for the reader to understand if they are reading a problem report from the early or late stages of the FW lifecycle.
For example; the Roland R-07 thread indicate a very flawed recorder (for our purposes) if you read reports in the early posts, but the R-07 evolved into a very capable unit in time when all FW updates were released - a FW version number here and there would ease the ability to navigate.
Aditionally; some reported consistent Bluetooth issues and others had no problem.
If reading through the entire thread you may extract that it seemed like iPhone users had few if any issues, while many with problems, if asked, seemed to be using Android. But most are statements like, "R-07 bluetooth sucks" vs. "I have no problems".
If more details were supplied, we may had been able to conclude if specific brands or models were more affected by connectivity problems.
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The clicking issue perhaps relating to stuff leaking from one board to another (sorry to be so technical) reminds me of the issue with the first production run of the Zoom M2 Mictrak device, which suffered serious RF breakthrough issues due to a production fault (according to Zoom). Zoom refunded customers and relaunched it some months later with the problem fixed. But the dire reviews on YouTube are still up, and I got mine for about half price (from the relaunch batch and it's fine). I really hope the slight problem with the AV2 isn't of that ilk. My thanks go to the brave early adopters, and thanks for the detailed testing and reporting!
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Fortunately these kind of weird ‘features’ seem like things that could be fixed by firmware updates. Not sure how Tascam responds to user feedback…
DR70D had lots of firmware updates as issues came to light.
I did a Better Business Bureau claim against TEAC America (Tascam parent company) because they advertised that the DR70D would run on 128GB cards, but then they said it can ONLY use cards from their list, which did not include any 128s like they had advertised.
They did not respond directly to my claim but did release a list with a working 128 card on it.
I can't find direct evidence of my claim online but I did make a thread about it.
https://taperssection.com/index.php?topic=175641.0 (https://taperssection.com/index.php?topic=175641.0)
Teac America's recent claim history:
https://www.bbb.org/us/ca/santa-fe-springs/profile/audio-visual-equipment/teac-america-inc-1216-40854/complaints (https://www.bbb.org/us/ca/santa-fe-springs/profile/audio-visual-equipment/teac-america-inc-1216-40854/complaints)
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Not sure how Tascam responds to user feedback…
Many years ago Tascam had a representative who engaged with us on the forum for a while. During the discussion Morst mentioned about the weird behavior and inconsistent info from the company he kinda threw up his hands and bailed. Several users got miffed that we "ran off" a company rep but he was just giving corporate standard non-answers to real users who were frustrated about failed recordings and lost opportunities. At one point he said that mountains of cards had been tested and approved but that info hadn't been updated on their website - I had to call them out for that as at the time I was running a small business single handedly and updating my website daily. If a global company like Tascam can't get their website updated with info that frustrated customers are clamoring for then there's a problem.
I have often used Tascam gear (DR2D, DR60D, DR701D, DR680) and will continue to. They make good, affordable prosumer level gear that appeals to me and I've had great success with their gear but their customer service is terrible.
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This ground has probably been covered extensively in other threads but just gotta say, damn, this being my first exposure to 32 bit float dual ADC recording, this shit is badass. It's awesome being able to just run DPA 4060s without a care in the world wrt levels. Would say the same for SBD patch but still gotta monitor at least a tiny bit to ensure I'm actually receiving the proper feed - fellow taper Roffels suggested I still run my A10 w/ the audio out from the AV2 so I don't have to sacrifice any Bluetooth range in that regard lol. So it'll just be glancing down, seeing levels, and then I'm set. Pretty amazing. Not like it was all that much overhead setting levels anymore, but just being totally free of it sure is some quality of life. The feed is just so CLEAN.
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fellow taper Roffels suggested I still run my A10 w/ the audio out from the AV2 so I don't have to sacrifice any Bluetooth range in that regard lol. So it'll just be glancing down, seeing levels, and then I'm set. Pretty amazing. Not like it was all that much overhead setting levels anymore, but just being totally free of it sure is some quality of life. The feed is just so CLEAN.
Good idea. If I owned an A10 I wouldn’t be buying the Bluetooth dongle. In fact it’s even making me consider buying a secondhand A10 instead of the AK-BT2 adapter.
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Curtis like its!
https://www.youtube.com/watch?v=w433J-C97Ec
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Curtis like its!
https://www.youtube.com/watch?v=w433J-C97Ec
Just watched this. It looks good I must admit and I think I will be purchasing when the price comes down a bit, and any initial bugs have been resolved.
He said 4 hours with one XLR mic using phantom power. This is much more in line with my expectations.
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Curtis like its!
https://www.youtube.com/watch?v=w433J-C97Ec
Curtis sums it up in the end; if you already have the F3 and it's working for you, keep using it. If you're looking at buying one of the two, he recommends the Tascam. Right now, I'd hold off until everything gets sorted out.
I'm using an F3 and F6 so I'll likely keep using those two as they work perfectly fine for my needs. I will admit, the iOS app for the Tascam (and its own display) both look better than the Zoom one but considering all I'm doing is locking the buttons on the F3 and controlling everything from the app, I really don't care all that much. Bluetooth didn't drain my Energizer Lithium Batteries all that much, either, so I'm glad to continue that as my workflow when I have zero issues walking into a venue.
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The FR-AV2 does not seem to have the high frequency hum lines (admittedly way out of hearing range) that the Portacapture X8 had, at least testing it with DPA4081 mics on 5V plug-in power.
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I commented on the 'Curtis' video that potential purchasers should read this thread to check on daily updates of real world end user experiences.
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I used the deck for Warren Haynes Band in Washington, D.C., last week.
https://archive.org/details/warrenhaynes2024-10-03gefell.vwmule.gefell
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Something that's still not clear to me about this recorder is the record level in 32bit float mode. I know there has been a lot of discussion and confusion about this (or should I say about the 'magnification') on the F3, and I think I understand what the F3 is doing.
I would expect that just like with the F3, a fixed gain is applied in the analog stage and then an amplification is applied in the digital domain depending on your record level setting. This would also mean that if the meters go over 0dB while recording, you can still fix that in post without problems.
However, if the record level does affect the analog gain, it would be possible to set the record level 'too high' or 'too low', introducing clipping or poor S/N.
Hopefully some of the proud new owners of the FR-AV2 can shed some light on this? I really hope it's a set-and-forget recorder where the record level doesn't really matter, apart from seeing the bars going nicely up and down on that beautiful display.
On a sidenote: If the record level doesn't really matter, than I think it's a bit odd that it's possible to change the record levels while recording. Doing so would only complicate things in post, I think...
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Something that's still not clear to me about this recorder is the record level in 32bit float mode. I know there has been a lot of discussion and confusion about this (or should I say about the 'magnification') on the F3, and I think I understand what the F3 is doing.
I would expect that just like with the F3, a fixed gain is applied in the analog stage and then an amplification is applied in the digital domain depending on your record level setting. This would also mean that if the meters go over 0dB while recording, you can still fix that in post without problems.
However, if the record level does affect the analog gain, it would be possible to set the record level 'too high' or 'too low', introducing clipping or poor S/N.
Hopefully some of the proud new owners of the FR-AV2 can shed some light on this? I really hope it's a set-and-forget recorder where the record level doesn't really matter, apart from seeing the bars going nicely up and down on that beautiful display.
On a sidenote: If the record level doesn't really matter, than I think it's a bit odd that it's possible to change the record levels while recording. Doing so would only complicate things in post, I think...
The technical aspect of audio is not my strong suit -- if you can give me a set of steps to test this properly I will do so.
With my limited understanding, I performed the following test:
Equipment:
DPA 4060 legacy > core sound BB > FR-AV2 EXT input
1) Identify a loud audio source that I know is not clipping at the mic level
2) record at +0 DB (showing -10 DB or so during recording)
3) record at +60 DB (showing endless clipping during recording)
4) normalize both files
No clipping in either once normalized. They both sound identical.
This means that the gain is not being applied at the analog level, right?
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I'd say yes, there's no analog gain.
Also, why would you use the core sound battery box with a recorder that provides 5v pip?
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I'd say yes, there's no analog gain.
Also, why would you use the core sound battery box with a recorder that provides 5v pip?
I thought 9v would provide maximal protection against distortion? Is there no difference between a 9v BB and 5v pip in that regard?
Edit: Hm. Did some further reading, I guess for 4060s it would seem not to matter? The answer would be "an abundance of caution", I always thought max allowable power was best. Thanks for pointing that out, I am still trying to discern just how certain it is.
https://taperssection.com/index.php?topic=190152.0#msg2299751
https://taperssection.com/index.php?topic=190550.msg2300230#msg2300230
Seeing conflicting info that makes me want to still run it to err on safe side..
https://taperssection.com/index.php?topic=187465.0
Don't wanna derail. But wanted to answer.
Ultimately I'm getting XLR adapters to get 9-10v out of the AV2 and cut the battery box out of the gear chain for my 406x's.
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Something that's still not clear to me about this recorder is the record level in 32bit float mode. I know there has been a lot of discussion and confusion about this (or should I say about the 'magnification') on the F3, and I think I understand what the F3 is doing.
I would expect that just like with the F3, a fixed gain is applied in the analog stage and then an amplification is applied in the digital domain depending on your record level setting. This would also mean that if the meters go over 0dB while recording, you can still fix that in post without problems.
However, if the record level does affect the analog gain, it would be possible to set the record level 'too high' or 'too low', introducing clipping or poor S/N.
Hopefully some of the proud new owners of the FR-AV2 can shed some light on this? I really hope it's a set-and-forget recorder where the record level doesn't really matter, apart from seeing the bars going nicely up and down on that beautiful display.
On a sidenote: If the record level doesn't really matter, than I think it's a bit odd that it's possible to change the record levels while recording. Doing so would only complicate things in post, I think...
The technical aspect of audio is not my strong suit -- if you can give me a set of steps to test this properly I will do so.
With my limited understanding, I performed the following test:
Equipment:
DPA 4060 legacy > core sound BB > FR-AV2 EXT input
1) Identify a loud audio source that I know is not clipping at the mic level
2) record at +0 DB (showing -10 DB or so during recording)
3) record at +60 DB (showing endless clipping during recording)
4) normalize both files
No clipping in either once normalized. They both sound identical.
This means that the gain is not being applied at the analog level, right?
Right! Thanks a bunch! :coolguy: This is exactly what I wanted to know and what I was hoping for. It means you can't set the level 'wrong'.
The only (not really relevant anymore) question remaining is what happens when the record level is changed while recording. Does it actually change the volume/amplitude of the samples stored in the file? I'd expect so. However, changing the record level while recording is something I would NEVER do with this device, as deemed useless by your previous test. IMHO it seems the F3 covers this functionality better by not allowing to change the record level while recording, only changing the vertical scale of the waveform display.
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Also, why would you use the core sound battery box with a recorder that provides 5v pip?
FYI:
I have a set of Core Sound Stealthy Cardioid Microphones and asked Len from Core Sound last week if I would be able to power these with the Tascam 5V PIP instead of the 9V battery box that came with the mics.
To my surprise I got a clear NO! Apparently the Core Sound battery box provides -9V, as apparently the mics need a negative voltage. The Tascam will provide +5V, so a positive voltage. I don't know the background of the need for this negative voltage, but he was very clear about this.
So I will still need the CS battery box when using these mics.
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My comment was about the dpas. I’m not sure what Len did with the Panasonic caps when building his, but needing negative voltage doesn’t make any sense to me.
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My comment was about the dpas. I’m not sure what Len did with the Panasonic caps when building his, but needing negative voltage doesn’t make any sense to me.
It doesn’t make sense to me either, but I have Len’s dpa binaurals and the possibility of plugging directly into the Tascam has me intrigued. His battery box is quite large, and made of metal which makes stealth unlikely.
My Sony D100 provides 5V+ PIP so I’ve just tried plugging directly into that and it doesn’t work. Probing with my voltage meter and I’m even more confused, but there would seem to be no path for voltage supply from the 3.5mm input to the mics.
If I’m going to use these mics with the Tascam it would seem that I’m going to need a female 4-pin Mini-XLR to TRS 3.5mm adapter. If anyone has any suggestions on where I can buy one I’d be grateful.
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My comment was about the dpas. I’m not sure what Len did with the Panasonic caps when building his, but needing negative voltage doesn’t make any sense to me.
It doesn’t make sense to me either, but I have Len’s dpa binaurals and the possibility of plugging directly into the Tascam has me intrigued. His battery box is quite large, and made of metal which makes stealth unlikely.
My Sony D100 provides 5V+ PIP so I’ve just tried plugging directly into that and it doesn’t work. Probing with my voltage meter and I’m even more confused, but there would seem to be no path for voltage supply from the 3.5mm input to the mics.
If I’m going to use these mics with the Tascam it would seem that I’m going to need a female 4-pin Mini-XLR to TRS 3.5mm adapter. If anyone has any suggestions on where I can buy one I’d be grateful.
I don't know what your CS battery box and mic wiring looks like, I've attached a photo of my CS mics and battery box. If I would want to supply 5V PIP to the mics, I'd only need to get a 4-pin mini-xlr female connector (so the same as in the battery box) and then short the two pin pairs, creating a direct path for each channel from the 3.5mm plug to the mics. Without this, there is indeed no connection between 3.5mm and mics (other than ground). But maybe your CS battery box is different, as mine provides this weird -9V voltage to the mics...
Example: https://www.thomann.de/nl/rean_tiny_xlr_rt4_mpr.htm (https://www.thomann.de/nl/rean_tiny_xlr_rt4_mpr.htm)
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For anyone curious, I did a "real world" (for me - batteries finished charging a few days ago, left in the AV2 since, used a few times here and there to fiddle with settings or minor <5m testing) battery test.
32bit/48khz/EXT input (stereo, no PIP).
Eneloop pros.
Got 8 hours 2 minutes out of it.
All files fine, including the last one which was cut off once the batteries died.
Max file size = 4,191,343 KB (splits after 3:06:16)
edit: Also, this discussion about -9v is interesting. I don't even know the model of this battery box, I bought it in like 2015 and can't find the receipt. Only ever used it once, just using it now because it fits the profile allowing for snug affixation to the AV2 on my rig. I set my rig up so it's all pre-built in my bag and able to be pulled out and set up on stage in less than 20 seconds.
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^ Is that with P48? What's the draw of the mics?
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^ Is that with P48? What's the draw of the mics?
No, I know it is likely not of particular use to most since people will probably run phantom or PIP. I will be running that test once I get my XLR adapters for the DPAs. Currently just testing exact circumstances of my upcoming outing this Saturday - so no phantom or PIP. Just BB powered mics into 3.5mm (EXT stereo).
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32bit/48khz/EXT input (stereo, no PIP).
Eneloop pros.
Got 8 hours 2 minutes out of it.
8 hours with external power equates to significantly less with phantom power or PIP.
Curtis in his review mentioned he got 4 hours with a single 48v microphone.
I suspected the claims made for the battery life in the early (paid for?) reviews would prove to be wildly optimistic. If we have to rely on external power options then it will really impact on the stealth possibilities.
The AV2 is on my radar to buy when it becomes available in the UK, but will resist the urge to buy until real life testing reports are in so appreciate your input Kyle K.
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32bit/48khz/EXT input (stereo, no PIP).
Eneloop pros.
Got 8 hours 2 minutes out of it.
8 hours with external power equates to significantly less with phantom power or PIP.
He used Eneloop Pro AA batteries, not an external battery. 8 hours on the Eneloops is outstanding! Add P48 and I bet he gets almost 4 hours. With an Anker external usb battery, P48 on, and it should drive it up to 8 hours.
https://www.bhphotovideo.com/c/product/1047694-REG/panasonic_bk_3hcca4ba_eneloop_xx_aa_rechargeable.html/?ap=y&ap=y&smp=y&smp=y&lsft=BI%3A514&gad_source=1&gclid=EAIaIQobChMIydOpjsmBiQMV-ahaBR3iGCqbEAQYAiABEgKHmPD_BwE
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32bit/48khz/EXT input (stereo, no PIP).
Eneloop pros.
Got 8 hours 2 minutes out of it.
8 hours with external power equates to significantly less with phantom power or PIP.
He used Eneloop Pro AA batteries, not an external battery. 8 hours on the Eneloops is outstanding! Add P48 and I bet he gets almost 4 hours. With an Anker external usb battery, P48 on, and it should drive it up to 8 hours.
https://www.bhphotovideo.com/c/product/1047694-REG/panasonic_bk_3hcca4ba_eneloop_xx_aa_rechargeable.html/?ap=y&ap=y&smp=y&smp=y&lsft=BI%3A514&gad_source=1&gclid=EAIaIQobChMIydOpjsmBiQMV-ahaBR3iGCqbEAQYAiABEgKHmPD_BwE
External mic power, not external recorder power
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External mic power, not external recorder power
Exactly.
I bet in the real world it ends up being two and a half hours with two 48v microphones connected.
Hopefully this device should be capable of sufficient hours with PIP mics for stealth situations.
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If I don't get home too late tonight (and assuming the eneloops are fully charged, I usually give them a full night and a day, threw them on the charger this morning) I'll test 48v recording time with Neumann km184.
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If I don't get home too late tonight (and assuming the eneloops are fully charged, I usually give them a full night and a day, threw them on the charger this morning) I'll test 48v recording time with Neumann km184.
Did you use the ‘power save’ function in your previous test?
(I know you can’t use it with phantom…)
Just curious ’cause the display is a potential power drainer…
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If I don't get home too late tonight (and assuming the eneloops are fully charged, I usually give them a full night and a day, threw them on the charger this morning) I'll test 48v recording time with Neumann km184.
That would be great. Thanks.
Battery life is one of the things that annoys me about some of the newer recorders because it seems to be getting worse, and I hate using external power.
I was disappointed at the amount of recording time I was getting with my Sound Devices MixPre-3 and bought the 8xAA sled, which still only took me to about three hours. That’s enough for my needs, but only just.
I did a test with my Sony XLR-1 48v power supply, which takes 4xAA batteries, connected two Sennheiser MKH-8040 mics and got 36 hours recording time, so I know what can be acheived it just seems that manufacturers aren’t concerned about battery life these days.
I have no experience with Zoom products though and some people have mentioned the excellent battery life here.
I just hope the Tascam FR-AV2 doesn’t disappoint like my Sound Devices did, so fingers crossed.
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Well damn, good news this time.
Gear:
Neumann KM184 (2x) > FR-AV2
Eneloop Pro (3x)
Settings:
MIC (Inputs 1 + 2)
48v
+0 DB
Backlight 5 Seconds
Backlight LOW brightness
CONTRAST set to minimum
Ni-MH under 'Battery'
(Everything else default)
Conditions:
Batteries sitting in recording deck for last 3 days after a full charge
~78 degree room
Turned on some tunes and placed the mics up against some headphones @ 50 mins to simulate recording conditions to a greater degree
(inconsistencies in waveform levels can be ignored, that's all me. everything sounds good.)
6 hours 15 minutes!
Very interesting / pleasantly surprising result.
Attached some pictures of the app throughout the process. When I saw the battery meter @ 3 hours I was thinking that the depletion rate would begin to accelerate rapidly, but it was relatively constant. Not perfectly accurate of course, didn't show half until about 3h45m (thought I took a screenshot @ half meter but I guess not) but accurate enough. Last screenshot is a warning it displays. Keeps the unit on with the last little bit of juice but automatically stops recording, and doesn't let you start it back up.
Re: TheJez, I was not using POWER SAVE when I did the EXT INPUT stereo (no PIP) battery test. I really should've. For that test, unless I experienced some sort of settings reversion glitch (checked after the fact), BACKLIGHT was set to ALWAYS @ MEDIUM brightness. Which is little odd because I don't remember setting that... must not have set it properly after my factory resets a few nights ago. I double checked 48v test settings prior to recording though, those are accurate.
Wouldn't have expected to get so much time given the ~8hrs with no phantom/PIP. Wonder how much juice was just the backlight setting vs the unit being surprisingly comparable between phantom & no phantom conditions.
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Thanks for that. This is very reassuring indeed, and puts the unit on my must buy list. Just wondering if I will own this instead of, or as well as, my MixPre-3 II since I only need two channels.
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Thanks for that. This is very reassuring indeed, and puts the unit on my must buy list. Just wondering if I will own this instead of, or as well as, my MixPre-3 II since I only need two channels.
My thought, too. I have a MixPre3 (not the ii) and feel like the FR-AV2 will be a better fit for what and how I record... Don't need both. Will purchase the Tascam and do a comparison with my mics, though, before letting go of the Sound Devices.
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I ran the deck last night for a show with DPA 4023. Used a 6,000 mah external battery and while the performance was just under an hour and a half, all four lights on the battery remained. So guessing very good run times with external battery.
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Re: TheJez, I was not using POWER SAVE when I did the EXT INPUT stereo (no PIP) battery test. I really should've.
Thanks! It's ok, it means we'll get even more runtime when POWER SAVE is enabled!
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Reading the manual I was very pleased to find there is a record hold function to prevent accidentally stopping a recording. Not a full key hold, but better than no hold at all.
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I ran the deck last night for a show with DPA 4023. Used a 6,000 mah external battery and while the performance was just under an hour and a half, all four lights on the battery remained. So guessing very good run times with external battery.
I’m absolutely not knowledgeable about batteries, so just out of curiosity I have this question: If you would have used the 2450mAh rechargeable AA batteries, wouldn’t you have had ‘more juice’ than when using this 6000mAh external battery? (As 3x2450=7350mAh) Or isn’t it that simple?
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The reports from Kyle (and other reports and reviews) made me bite the bullet: I went for the FR-AV2 instead of the F3... Should arrive tomorrow :yahoo:. Thank you all for your reports, info and valuable opinions!
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I ran the deck last night for a show with DPA 4023. Used a 6,000 mah external battery and while the performance was just under an hour and a half, all four lights on the battery remained. So guessing very good run times with external battery.
I’m absolutely not knowledgeable about batteries, so just out of curiosity I have this question: If you would have used the 2450mAh rechargeable AA batteries, wouldn’t you have had ‘more juice’ than when using this 6000mAh external battery? (As 3x2450=7350mAh) Or isn’t it that simple?
The batteries are likely in series, which increases voltage, rather than parallel which increases capacity.
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Before and after photos.
Modifying my Core-Sound DPA4061 microphones ready for the AV2 with 5v PIP. (With apologies to Len).
Both mics now terminate in a single 3.5mm TRS jack. I have tested with my Sony PCM-D100, which also provides 5v PIP, and it works great.
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Happy to be of help testing. It's what I do for a living :)
On the clock-bleed-in-extreme-edge-cases front, received a response back from TASCAM:
Hello Kyle,
We have forwarded the information to our engineering department in Japan.
They were able to duplicate the issue and are working on a firmware update which should be available in the next few weeks.
Thank you for your interest in TASCAM products.
When I reached out to them I acknowledged this could be expected functionality, I just asked them to see if it was reproducible on a test unit & to rule out some sort of manufacturing issue with shielding.
Kinda interesting - I wonder if the fix will be to simply reduce the max gain able to be applied since it's mainly just scary to see on the display. It is difficult to conceive of a valid use case where the clicks would be at all perceptible (outside of perhaps surreptitious eavesdropping, but even then you'd probably be using a higher sensitivity mic where it wouldn't be an issue haha).
Just updating to bookend that whole thing. Er, I guess the true bookend won't come till the firmware update does..
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Excellent work, Kyle. With recording equipment one can take the approach that minor imperfections won't be heard anyway, or that if it can be perfect, it should be perfect, audible in real world or not. But it's usually the case that what makes the biggest difference is the selection and placement of mics. I guess before long someone will bring out an AI tool to deal with that...
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Out in the field for the first time running 2 AV2s. In a medium-small room (Mr Smalls Pittsburgh, about 30-40 ft from SBD AV2) and yeah - PCM A10 does indeed smoke it on the Bluetooth front. Totally stable, while there's no connecting to that AV2. Saving me from feed anxiety (just wrt signal, not levels). Really the only damper for me, otherwise love it.
Edit: note this is Android, tested on my older S10+ as well as my wife's flagship model (S23 ultra or whatever newest is). It is very well possible that iPhone would have a better experience given what are often dramatically different development processes.
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I got mine a couple of days ago. Just fooled around with at home since I don't have any shows coming up right away due to lack of free time.
First impressions - very light but it feels solid, I like the display and all the info I could want is there, the control interface is going to take some getting used to (and I would prefer physical dedicated switches for some things that are menu controlled but I get why they did it the way they did to save space that switches require) but any time I get a new recorder there's some learning curve getting accustomed to it, battery life is sufficient using well worn Eneloop AAs to get me through a opener and headliner set on one set of 3 batteries using my MBHO actives.
Looking forward to getting out to some shows with it. Due to opening a new bar and restaurant this week and my hair being basically on fire I may not get a chance to use it until Bob Weir and Wolf Bros with the Louisiana Philharmonic in New Orleans in November.
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Got mine yesterday. Also bought and charged a bunch of Ikea Ladda 2450mAh batteries as these get good reviews. Also got a SanDisk Extreme Pro microSDHC card of 32GB, rated 90MB/s write and 100MB/s read speed.
First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
- The rubber microSD-card 'door' feels rather flimsy. Will probably come off after a couple of years unless you'll never use it.
Did a duration test with freshly charged Ladda batteries, 48kHz 32bit float, +5V PIP for a pair of Clippy microphones through the EXT input, using display setting as low as possible:
- Auto power save: OFF
- Power save mode: OFF
- Backlight: OFF
- Brightness: LOW
- Indicators: ALL OFF
- Contrast: as low as possible
This gave me a rather dark but still readable display, and... A whopping 9:13 of recording time!!
But:... After checking the recorder about 1.5h into the test, I found that the VU meters of both input and output had frozen in time. The recording time was still running, I could still add markers and change the headphone output volume (both confirmed on the display) and after the test I verified that it had been recording the whole time. This partly freezing display (or more likely crashed VU metering software) is not so good...
After putting in new batteries and switching on the device, the VU meters are fine agian. Now repeating the test with display settings as high as possible, just curious what will come out of that and if the VU meters will freeze again...
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I'll be keeping my eye out for those Mixpre-3's ya'll are getting rid of!
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I'll be keeping my eye out for those Mixpre-3's ya'll are getting rid of!
For a brief moment I was thinking I would replace mine with the Tascam FR-AV2 because there are a few things about my Mixpre-3 II that I don’t like:
The battery life
Only 2.5v PIP
The fact that plugging in external 3.5mm microphones doesn’t use the Kashmir preamps
It’s a bit of a pain to change the input to AUX
I was thinking about getting the Earsight PIP/XLR Adapter for €45 which would have solved the last three issues, but now thinking of getting the AV2 in addition to the Mixpre-3 and using it mainly as a dedicated 3.5mm microphone recorder and for stealth.
What the Mixpre-3 does have is fantastic preamps, and paired with my Sennheiser MKH8040 mics I have made some of my favourite ambient recordings. Also the form factor of the Mixpre-3 with mounting points top and bottom is very useful.
If my Mixpre-3 II is ever to find its way into the classifieds the Tascam AV2 is going to have to produce low noise recordings just as good. I doubt that will happen but look forward to comparisons.
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hm, looks like I'll have another ticket for TASCAM.
What's odd is I didn't notice this in my initial battery tests, I could've sworn I normalized the split tracks during that process but maybe not.
What's attached here is zoomed in waveform at the end of the automatic file split. This is present in both devices I was using. Audible as a tiny "pop" of distortion, normalizes to -INF. When it's simply cut out it is barely noticeable in the recording, but still, should not be occurring.
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hm, looks like I'll have another ticket for TASCAM.
What's odd is I didn't notice this in my initial battery tests, I could've sworn I normalized the split tracks during that process but maybe not.
What's attached here is zoomed in waveform at the end of the automatic file split. This is present in both devices I was using. Audible as a tiny "pop" of distortion, normalizes to -INF. When it's simply cut out it is barely noticeable in the recording, but still, should not be occurring.
What software are you using? Could it be non-sample data at the end of the file being interpreted as samples?
I will check my test recording tomorrow.
How do you report issues to TASCAM? I’m having major problems changing the record level in the app…
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hm, looks like I'll have another ticket for TASCAM.
What's odd is I didn't notice this in my initial battery tests, I could've sworn I normalized the split tracks during that process but maybe not.
What's attached here is zoomed in waveform at the end of the automatic file split. This is present in both devices I was using. Audible as a tiny "pop" of distortion, normalizes to -INF. When it's simply cut out it is barely noticeable in the recording, but still, should not be occurring.
You're on to something. See attachment for an automatic file split on my recorder after in Adobe Audition. It seems a new file always starts from silence. Not a seemless transition as one would expect.
Recorder FW 1.01...
I also noticed way more markers than expected when enabling 'time marker' every 10 minutes, e.g. 83 markers at 0:00:00.000...
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First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
Thanks for reporting your findings. How easy is it for you change the levels during recording?
Regarding the transfer speed - this is something I was wondering about when the specs were posted before the release.
The recorder has USB-C but only USB 2.0 transfer speeds. I don't understand that decision by Tascam...
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First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
Thanks for reporting your findings. How easy is it for you change the levels during recording?
Ehm, well, it's not easy as with dedicated buttons, obviously. Assuming you enabled 'Gang', which links the record level of channel 1 and 2, you need to do this: Press Input button (F3) -> Enter (F4) to select Channel 1. Then Press F2 or F3 to decrease/increase the record level, then press Enter (F4) when done and Home (F1) to return to the Home screen.
However: When recording in 32bit float, it doesn't make any sense to change the record level while recording. The analog part is set to a fixed gain, and any change in record level is handled in the digital domain which could/should/very likely will be done in post anyway. It will only complicate things in post if you'd change the record level during recording, as it will affect the amplitude of the samples stored in the file. The only reasons I could think of are a) you may get better levels when playing back the recording from the device or b) the vu-meters will show the input better scaled if you like that. In my use cases, neither seem a valid reason to change the levels while recording, but this may be different for others. I'd try to set the levels sort of best-guess right before hitting 'record', and then I'll never tocuh it even when the levels are very low or even clipping. Both will be fixed during post without quality loss thanks to the dual-ADC and 32bit float recording.
One additon: When using the App (at least on iPhone), the record level control is rather cumbersome. When 'gang' is not enabled, and then changing the level on channel 1, then channel 2 often suddenly drops to 1dB (in range of 0..60dB)!! And vice versa (channel 1 dropping to 1dB when controlling channel 2 record level). When 'gang' is enabled, controlling isn't very smooth either. E.g. when sliding from 20 to 30dB, the sliders will lag behind and go from 20 to 30 over a couple of seconds, and while doing so occasionally drop to 1dB for brief moments. These hickups are not just visual, but really affect the recording. I guess another reason not to change record level while recording :-\
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First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
Regarding the transfer speed - this is something I was wondering about when the specs were posted before the release.
The recorder has USB-C but only USB 2.0 transfer speeds. I don't understand that decision by Tascam...
Well, as someone with a bit of experience in consumer electronics, I can imagine a bit what's going on. The USB interface on this device is not a dedicated disk interface. It can act in different modes, in this case as a disk interface and as a sound card. The choice between the two modes and the associated processing is done in the software, which means it is software that is 'glueing' the microSD card to the USB bus when the device is set to 'SDCard Reader Mode'. Each block read/written is therefor likely passed through the software, which will make it slower than more hardware-oriented solutions. It's a real pitty, especially since the microSD 'door' on the device is flimsy...
I think most recorders will act this way.
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I also noticed way more markers than expected when enabling 'time marker' every 10 minutes, e.g. 83 markers at 0:00:00.000...
It seems that each file gets ~100 markers at 00:00:00.000 'for free' right from the start. It seems they are filled in during recording, e.g. with the time marks, manual marks or the 'peak mark'.
I can imagine they are doing it this way, as the markers are stored at the start of a WAV file. They just claim space for ~100 markers at the moment a file is created and then start appending the samples to it. If a marker must be stored, they just update an existing 00:00:00.000 marker with the right timestamp and type/name.
Maybe the number of claimed markers depend on the sample rate, I haven't tested that.
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Got mine yesterday. Also bought and charged a bunch of Ikea Ladda 2450mAh batteries as these get good reviews. Also got a SanDisk Extreme Pro microSDHC card of 32GB, rated 90MB/s write and 100MB/s read speed.
First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
- The rubber microSD-card 'door' feels rather flimsy. Will probably come off after a couple of years unless you'll never use it.
Did a duration test with freshly charged Ladda batteries, 48kHz 32bit float, +5V PIP for a pair of Clippy microphones through the EXT input, using display setting as low as possible:
- Auto power save: OFF
- Power save mode: OFF
- Backlight: OFF
- Brightness: LOW
- Indicators: ALL OFF
- Contrast: as low as possible
This gave me a rather dark but still readable display, and... A whopping 9:13 of recording time!!
But:... After checking the recorder about 1.5h into the test, I found that the VU meters of both input and output had frozen in time. The recording time was still running, I could still add markers and change the headphone output volume (both confirmed on the display) and after the test I verified that it had been recording the whole time. This partly freezing display (or more likely crashed VU metering software) is not so good...
After putting in new batteries and switching on the device, the VU meters are fine agian. Now repeating the test with display settings as high as possible, just curious what will come out of that and if the VU meters will freeze again...
Result of repeated test with same settings, except that all display settings are now set as high as possible: 7:34 recording time. So the display is a power-hungry thing, something to consider especially when also using phantom power... Oh, and no freezing VU-meters this time...
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First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
Thanks for reporting your findings. How easy is it for you change the levels during recording?
When recording in 32bit float, it doesn't make any sense to change the record level while recording. The analog part is set to a fixed gain, and any change in record level is handled in the digital domain which could/should/very likely will be done in post anyway.
My idea and hope was to be able to use the recorder as a two channel mixer. If you're interviewing two people on two separate mics on channels 1 and 2, you could lower the levels during recording to prevent the need to do so much post work. However, it seems to be a set it and work it in post mentality. There is always the possibility of putting a small mixer between the recorder and the mics, though.
There are not many two track recorders that actually let you adjust individual levels. The Fostex FR-2LE comes to mind, of course that's a legacy machine these days and records to CF cards!
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One additon: When using the App (at least on iPhone), the record level control is rather cumbersome. When 'gang' is not enabled, and then changing the level on channel 1, then channel 2 often suddenly drops to 1dB (in range of 0..60dB)!! And vice versa (channel 1 dropping to 1dB when controlling channel 2 record level). When 'gang' is enabled, controlling isn't very smooth either. E.g. when sliding from 20 to 30dB, the sliders will lag behind and go from 20 to 30 over a couple of seconds, and while doing so occasionally drop to 1dB for brief moments. These hickups are not just visual, but really affect the recording. I guess another reason not to change record level while recording :-\
Are you going to report this to tascam to see if it can be improved?
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First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
Thanks for reporting your findings. How easy is it for you change the levels during recording?
When recording in 32bit float, it doesn't make any sense to change the record level while recording. The analog part is set to a fixed gain, and any change in record level is handled in the digital domain which could/should/very likely will be done in post anyway.
My idea and hope was to be able to use the recorder as a two channel mixer. If you're interviewing two people on two separate mics on channels 1 and 2, you could lower the levels during recording to prevent the need to do so much post work. However, it seems to be a set it and work it in post mentality. There is always the possibility of putting a small mixer between the recorder and the mics, though.
There are not many two track recorders that actually let you adjust individual levels. The Fostex FR-2LE comes to mind, of course that's a legacy machine these days and records to CF cards!
Oh, that’s absolutely possible! If you disable the ‘gang’ option, you can set the record level for each channel individually! No need for external mixer…
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One additon: When using the App (at least on iPhone), the record level control is rather cumbersome. When 'gang' is not enabled, and then changing the level on channel 1, then channel 2 often suddenly drops to 1dB (in range of 0..60dB)!! And vice versa (channel 1 dropping to 1dB when controlling channel 2 record level). When 'gang' is enabled, controlling isn't very smooth either. E.g. when sliding from 20 to 30dB, the sliders will lag behind and go from 20 to 30 over a couple of seconds, and while doing so occasionally drop to 1dB for brief moments. These hickups are not just visual, but really affect the recording. I guess another reason not to change record level while recording :-\
Are you going to report this to tascam to see if it can be improved?
That is my intention. I’ll try to figure out tomorrow how to do that. Any suggestions?
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hm, looks like I'll have another ticket for TASCAM.
What's odd is I didn't notice this in my initial battery tests, I could've sworn I normalized the split tracks during that process but maybe not.
What's attached here is zoomed in waveform at the end of the automatic file split. This is present in both devices I was using. Audible as a tiny "pop" of distortion, normalizes to -INF. When it's simply cut out it is barely noticeable in the recording, but still, should not be occurring.
You're on to something. See attachment for an automatic file split on my recorder after in Adobe Audition. It seems a new file always starts from silence. Not a seemless transition as one would expect.
Recorder FW 1.01...
I also noticed way more markers than expected when enabling 'time marker' every 10 minutes, e.g. 83 markers at 0:00:00.000...
Hmm, so you do not see distortion at the end of the file(s)? Interesting. Either way it doesn't seem perfectly smooth as would be preferable of course.
I went through their support site:
https://tascam.com/us/support/contact
There is an icon that opens a page which allows you to select device model, FR-AV2 is on the list. Then it will open a ticket with their support team. I work in product development so I have experience 'speaking their language', just a matter of being both succinct and comprehensive (where relevant) with extremely clear steps to reproduce, and boiling it down to one specific question if possible.
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Hmm, so you do not see distortion at the end of the file(s)? Interesting. Either way it doesn't seem perfectly smooth as would be preferable of course.
I went through their support site:
https://tascam.com/us/support/contact
There is an icon that opens a page which allows you to select device model, FR-AV2 is on the list. Then it will open a ticket with their support team. I work in product development so I have experience 'speaking their language', just a matter of being both succinct and comprehensive (where relevant) with extremely clear steps to reproduce, and boiling it down to one specific question if possible.
No, I don't have any distortion at the end of the file(s), only a problem at the start of the new file(s). I noticed the same issue when doing manual split.
Thanks for pointing me to the support page. I've reported two issues in separate tickets: 1) Non-seamless file splits and 2) erratic record level control from the App...
Both reports resulted in an interesting mail response: "thank you for contacting TASCAM. Our offices are now closed for the weekend. Your message will be read, when we return on monday."
It's Tuesday today, isn't it? ???
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Yesterday was a U.S. holiday. I still haven't received my direct deposit >:(
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Yesterday was a U.S. holiday. I still haven't received my direct deposit >:(
Ah, that explains… Thanks :facepalm:
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On paper, this looks really great and may be a better alternative to the Zoom F3.
I am most interested to see how its autoranging ADCs perform, and if they suffer from the same windowing noise modulation as the Zoom F-series does.
I have seen you mention this "windowing noise modulation" on the F-series in a couple places, but I don't fully understand what is being discussed / the issue here. Could you elaborate a bit on how this is an issue and what different outcomes can be achieved with other 32-bit devices that don't suffer from this problem (e.g. SoundDevices MPx-ii series)? Thanks in advance for the education!
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^ The discussion starts here: https://taperssection.com/index.php?topic=192100.msg2409825#msg2409825 (https://taperssection.com/index.php?topic=192100.msg2409825#msg2409825). It seems like the original YouTube is no longer publicly available, though.
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Based on my first hand experiences, I would say that this "problem" with the F3 is purely theoretical and completely inaudible for our purposes,
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On paper, this looks really great and may be a better alternative to the Zoom F3.
I am most interested to see how its autoranging ADCs perform, and if they suffer from the same windowing noise modulation as the Zoom F-series does.
I have seen you mention this "windowing noise modulation" on the F-series in a couple places, but I don't fully understand what is being discussed / the issue here. Could you elaborate a bit on how this is an issue and what different outcomes can be achieved with other 32-bit devices that don't suffer from this problem (e.g. SoundDevices MPx-ii series)? Thanks in advance for the education!
When you're recording very quiet sounds (and the more sensitive ADC is presumably engaged), and then a sudden louder sound engages the less sensitive ADC, the transients will contain the higher noise floor of the less sensitive ADC. Imho the loudness of the sound will do a good job masking that noise, but some sound designers have complained about this because they can't profile that noise for software noise reduction (because it's only present during the transients). I was able to hear the noise on the video that was taken down when my attention was drawn to it, but again, it is not something I'd worry about. I've never actually heard it in the live music I record.
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New firmware out, fixing the volume issue I am experiencing while using the APP over BT
https://tascam.com/us/product/fr-av2/support
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New firmware out, fixing the volume issue I am experiencing while using the APP over BT
https://tascam.com/us/product/fr-av2/support
Great! And apparently also the 1Hz ticks as reported by Kyle should be fixed!
Next up hopefully the non-seamless splits… :headphones:
Edit: Just confirmed that indeed volume control from App and 1Hz ticks have been solved with FW1.02. File split still not seamless :-\
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There are not many two track recorders that actually let you adjust individual levels. The Fostex FR-2LE comes to mind, of course that's a legacy machine these days and records to CF cards!
The TASCAM DR100 MK3 lets you adjust individual levels. You choose L or R or both inputs by means of a physical switch on the front.
That unit is also one of the few with a digital input.
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And it’s digitally controlled in 0.5dB steps
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Another interesting battery test, not sure if it will be useful to anyone else but just incase: (TLDR seemingly more time out of 24v XLRs vs EXT for PIP mics, more testing to come)
Picked up this Dual XLR > 3.5mm phantom power converter (I couldn't discern solid consensus on if it was optimal to run DPA 406X's with 9v vs 5v PIP and want to play it safe - plus the mic inputs are rated at a higher max input, though it is a bit unlikely to ACTUALLY come into play..)
https://immersivesoundscapes.com/accessories/Adapter-for-PIP-mics-to-work-on-XLR-48V-Recorder-p492004466
Set inputs to 24v phantom power, otherwise same settings as before wrt 5 seconds backlight and low contrast 32bit etc, Eneloop Pros
I had 7.5 hours left on my SD card and didn't expect it to make it this far given the ~8h of recording time on PIP. But it made it the full 7.5h and I was left with what appears to be a good amount of juice. Screenshot attached is battery meter after recording for 7.5h and running for 9h total (supplying power to mics the entire time!)
I will wipe my card and do a full length test but this is indicating that I'd actually get more time out of the converted phantom power mic inputs compared to EXT INPUT PIP?? That seems odd, right? Not that I'm complaining. I should be running these tests multiple times and averaging the results for proper analysis, too - wonder if my og PIP recording test could have been an outlier and if I'd get more time with repeated testing. Just seems so odd to get more time out of dual 24v.
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Congratulations to those here whose feedback has prompted the FW update, and kudos to Tascam for quick fixes.
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Got mine yesterday. Also bought and charged a bunch of Ikea Ladda 2450mAh batteries as these get good reviews. Also got a SanDisk Extreme Pro microSDHC card of 32GB, rated 90MB/s write and 100MB/s read speed.
First findings:
- The USB file transfer speed is rather disappointing (as with most recorders): Just about 5.5MB/s. Getting your recording off the device this way will take forever. When the same microSD card is put in an El Cheapo USB3 hub with card reader, I get 30MB/s.
Just did one more test: I put the 32GB SanDisk Extreme Pro microSDHC card directly into the SD-card reader in my DELL laptop, and get 80MB/s write speed and 90MB/s read speeds. That makes the 5.5MB/s I get when using the recorder as SD-card reader even more disappointing...
To put things into perspective: With 32bit/48kHz, a ~3-hour recording takes up ~4GB. Transferring ~4GB at 5.5MB/s takes ~12.5 minutes. Seems like a good moment to grab a cup of coffee and go to the toilet.
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Congratulations to those here whose feedback has prompted the FW update, and kudos to Tascam for quick fixes.
I think most credits go to Kyle. I reported my findings just three days ago, and all I got so far is an automated message 'it's weekend now, we'll read it on Monday'. I guess the solved issues have been 'known issues' for a bit longer.
I really hope they will be able to address the file split issue, although this only becomes a real problem when recording longer shows (>3h at 32bit/48kHz) or when using higher sample rates (>1.5h at 32bit/96kHz) or when using manual split...
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Congratulations to those here whose feedback has prompted the FW update, and kudos to Tascam for quick fixes.
I think most credits go to Kyle. I reported my findings just three days ago, and all I got so far is an automated message 'it's weekend now, we'll read it on Monday'. I guess the solved issues have been 'known issues' for a bit longer.
I really hope they will be able to address the file split issue, although this only becomes a real problem when recording longer shows (>3h at 32bit/48kHz) or when using higher sample rates (>1.5h at 32bit/96kHz) or when using manual split...
Even though I do it for a living at a product-based company (capturing, reporting, & validating defects at various stages of release) it is immensely more satisfying doing it on my own time haha. So it is cool to see. Though it's funny that the release notes made it sound as though it was solely an issue with no mics connected, which wasn't the case :P Probably a PO writing based upon the original repro steps in the ticket. Ultimately just glad to help.
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I can't find the actual firmware on that page. Has it dropped or just the notice? Very confusing page.
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Has anyone used this with lavalier mics yet? Can you confirm if they are automatically detected, or does the input type have to be chosen in the menu?
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I can't find the actual firmware on that page. Has it dropped or just the notice? Very confusing page.
Yeah, I don't see the actual FW releases anymore either. Weird... Fortunately I still have the downloaded file. See attachment.
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Has anyone used this with lavalier mics yet? Can you confirm if they are automatically detected, or does the input type have to be chosen in the menu?
You need to set the input source yourself in the menu. Options are:
- Mic (=XLR)
- Line (=TRS)
- EXT(ST) (=3.5mm)
- EXT(MONO) (=3.5mm)
- USB
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I can't find the actual firmware on that page. Has it dropped or just the notice? Very confusing page.
Yeah, I don't see the actual FW releases anymore either. Weird... Fortunately I still have the downloaded file. See attachment.
Thank you
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I can't find the actual firmware on that page. Has it dropped or just the notice? Very confusing page.
Yeah, I don't see the actual FW releases anymore either. Weird... Fortunately I still have the downloaded file. See attachment.
That likely means another update is almost ready to be released.
It is a shame that companies these days expect their customers to be their beta testers (except for Sony, perhaps). The non-seamless splits are scandalous...
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This device was announced well after some recent Zoom announcements but released well before. Rushed out, or maybe they don't announce way in advance like Zoom? Hard to know...
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I can't find the actual firmware on that page. Has it dropped or just the notice? Very confusing page.
The firmware is back online! 1.02 is still the latest…
https://tascam.com/us/product/fr-av2/support (https://tascam.com/us/product/fr-av2/support)
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I can't find the actual firmware on that page. Has it dropped or just the notice? Very confusing page.
Yeah, I don't see the actual FW releases anymore either. Weird... Fortunately I still have the downloaded file. See attachment.
That likely means another update is almost ready to be released.
It is a shame that companies these days expect their customers to be their beta testers (except for Sony, perhaps). The non-seamless splits are scandalous...
I couldn't agree more, and I'm baffled by the people who think the distorted file splits are no big deal. That's something pretty much every field recorder released for well over 20 years can do without a hitch.
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I'm baffled by the people who think the distorted file splits are no big deal.
I guess this remark is addressed to my remark about >3h file splits, as I don't see any other post that this might be referring to. I never said this is not a big deal, though! I totally agree that seamless file splits are a very basic requirement and it is a shame it can't do it properly right now. That's also why I reported this issue to Tascam.
However, one's only affected by the file split bug if one's actually splitting files, right? Either automatically at 4GB boundaries or manually. I see no problem in making owners of this device aware of this, so they can, at the moment, try to avoid a hickup at some random point in their recording by preventing recording >3h@48kHz at once (e.g. by stopping/restarting between support act and main act) or by not using the manual split feature. Until a fix for this bug becomes available, of course. If you think you may have to record something flawlessly >3h@48kHz, or >1.5h@96kHz, then this recorder is not the right device for you at the moment. For me personally, I can't think of any recording I made in the past 35 years that lasted more than 3h . (YMMV. I must admit I never taped Bruce Sprinsteen or other artists that tend to play on and on and on... 8) )
I'm convinced they're working on it... They can't afford not to fix this.
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or me personally, I can't think of any recording I made in the past 35 years that lasted more than 3h . (YMMV. I must admit I never taped Bruce Sprinsteen or other artists that tend to play on and on and on... 8) )
I'm convinced they're working on it... They can't afford not to fix this.
I've done quite a few over the years - including days where I used a data grade DATs that I still needed to flip somewhere before the 3 hour mark. With Digital, there have been days where I'd rather just set it and forget it at a stage so I can be somewhere else, so there are use cases out there where this will apply but it's not the norm.
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or me personally, I can't think of any recording I made in the past 35 years that lasted more than 3h . (YMMV. I must admit I never taped Bruce Sprinsteen or other artists that tend to play on and on and on... 8) )
I'm convinced they're working on it... They can't afford not to fix this.
I've done quite a few over the years - including days where I used a data grade DATs that I still needed to flip somewhere before the 3 hour mark. With Digital, there have been days where I'd rather just set it and forget it at a stage so I can be somewhere else, so there are use cases out there where this will apply but it's not the norm.
This is me, regularly. Especially given the short bluetooth range, it is a pain stopping and starting both the stage rig & SBD/FOB setup(s) given that I have to get to/from the front of packed venues.
Working with the imperfect splits wasn't too bad but it was definitely a major disappointment and ridiculous for it to have made it to production. I could go on at length regarding the depressing nature of modern quality practices but at this point it's just screaming into the void.
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Working with the imperfect splits wasn't too bad but it was definitely a major disappointment and ridiculous for it to have made it to production. I could go on at length regarding the depressing nature of modern quality practices but at this point it's just screaming into the void.
Too many marketing graduates and not enough tech guys at TASCAM
Already too many engineers..
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If you think you may have to record something flawlessly >3h@48kHz, or >1.5h@96kHz, then this recorder is not the right device for you at the moment.
It is fine as long as you have TWO of them! Start 'em ten minutes apart or whatever, fix in post.
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If you think you may have to record something flawlessly >3h@48kHz, or >1.5h@96kHz, then this recorder is not the right device for you at the moment.
It is fine as long as you have TWO of them! Start 'em ten minutes apart or whatever, fix in post.
Or just stick with an F3 or buy one of those if this is important to you. I know that unit has no noticeable issues.
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Perhaps my perspective on long running times may be unusual, but 32 bit float does give rise to a new possibility when recording live events (openly, or by request). It seems to me that it's now possible to set up an appropriately located stereo mic system on a stand, connected to a 32 bit float recorder at the bottom. Turn it on an walk away. No need to run mic cables to a remote control room, taped down to avoid tripping hazards, and all that stuff. Actually, using something like the Zoom M3 MicTrak lends itself even more to that way of working. In the classical music scene one might well record a rehearsal and then a concert, and so you'd leave the rig running (from a USB power bank) for several hours. So glitch free file splits would be kind of essential.
I've not actually done that in this century, but in the last century I did record three classical gigs at once in three different halls in a broadly similar fashion. But the level setting had to be based on years of experience, and conservative!
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Just a warning to everyone!
I attached an external powerbank to the Tascam FR-AV2 USB port via a usb C cable to usb C from a SmallRig powerbank.
Result: Instant death of the Tascam. It will not start or respond.
Any suggestions?
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Can you tell us more about the power bank? Was it a high power / fast charge type or something 'ordinary'? Was the Tascam powered on when you connected it? (I'm sorry to hear your adverse experience!)
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It was a SMALLRIG NP-F970 10500mAh Replacement Battery for Sony NP-F970. I have three of them
- they come in different colours. I have used them as a NP-970 replacement on my SoundDevices MixPre recorders (3 and 6) very succesfully.
They have USB2 (out) and USB3 (in and out) ports. I have also used them as charging devices on pretty much anything with USB 3 in - iPads, phones, Zoom F3 and so on. They have worked beautifully and reliably for me and they have a digital percentage remaining capacity display.
I can't remember whether the Tascam was on or off when plugging in the USB 3 cable - I think it may have been on (?). The cable had USB 3 connections at each end - as supplied by Tascam.
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Just a warning to everyone!
I attached an external powerbank to the Tascam FR-AV2 USB port via a usb C cable to usb C from a SmallRig powerbank.
Result: Instant death of the Tascam. It will not start or respond.
Any suggestions?
Wow, that's not good! I connected an Anker power bank earlier without issues.
Did you contact Tascam support? Or your vendor? I really hope you'll be able to get it returned/exchanged.
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I wish all of this was surprising. If Tascam sent a handful of devices to the early adopters here to test before rollout, they'd save a fortune in the long run.
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Just a warning to everyone!
I attached an external powerbank to the Tascam FR-AV2 USB port via a usb C cable to usb C from a SmallRig powerbank.
Result: Instant death of the Tascam. It will not start or respond.
Any suggestions?
Are you sure that usb C connection on the powerbank is also an output? On my powerbanks all usb C are input only. The output is provided through a usb A type connector.
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Thanks to all who have commented.
Joop:
The USB C output is labelled 'Input/Output'; the USB A output is labelled 'Output'.
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Just a warning to everyone!
I attached an external powerbank to the Tascam FR-AV2 USB port via a usb C cable to usb C from a SmallRig powerbank.
Result: Instant death of the Tascam. It will not start or respond.
Any suggestions?
Did you try a fresh set of AA batteries?
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Here's a short update regarding the file split issues. I still didn't get any another response from Tascam U.S.A. Support than "It's weekend, we'll read your message on Monday", so I put in another report, this time with Tascam Europe. Maybe the USA branch ditched my report when they found out my device was sold in Europe, I don't know. At least now I got an (automated) message saying they'll be in touch. We'll see.
Anyway, I was curious if there was more to say to this issue, rather than 'file split not seamless', so I generated a 400Hz test tone and played it on my stereo, then recorded it with the Tascam and manually generated a couple of file splits. After checking each split in my editor, it seems that the file splits are seamless in time, but not seamless in amplitude. It's just as if 'someone' suddenly turns down the volume at the split, and then gradually turn it back to the right level over a period of 3-4ms.
Just FYI.
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Hmm. Could do better! I will add that test to those on my Zoom H1 XLR when it arrives tomorrow. Obviously not a direct rival device but has some broad similarities. Well, you know, it makes recordings... :facepalm:
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It was a SMALLRIG NP-F970 10500mAh Replacement Battery for Sony NP-F970. I have three of them
I find your device very confusing, downloaded the manual and found out that it does have a very high output at the 5 Volt section of 8.4 Volt, this might be too much. As it is in English and Chinese with both same values I don't think its a misprint.
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It was a SMALLRIG NP-F970 10500mAh Replacement Battery for Sony NP-F970. I have three of them
I find your device very confusing, downloaded the manual and found out that it does have a very high output at the 5 Volt section of 8.4 Volt, this might be too much. As it is in English and Chinese with both same values I don't think its a misprint.
Sounds like the SMALLRIG NP-F970 is very much not a standard "USB power bank" so using it in applications other than what it was designed for (cameras) is quite risky.
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Just updated version 102 . last night i had some dropouts on channel 1 First set fine second set about 6 min mar channel 1 had intermittent drop offs then ok then drop offs it finally cleared up about the 8min mark of the second set . last night was with version 1.00 tried to duplicate all good . had DPA 4080's xlr in Linked and ganged . maybe the new firmware will fix this but it maybe something else 128 g Sandisk ultra should be fin e't know what the issue was .
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SMALLRIG NP-F970 << Not a power bank.
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Just updated version 102 . last night i had some dropouts on channel 1 First set fine second set about 6 min mar channel 1 had intermittent drop offs then ok then drop offs it finally cleared up about the 8min mark of the second set . last night was with version 1.00 tried to duplicate all good . had DPA 4080's xlr in Linked and ganged . maybe the new firmware will fix this but it maybe something else 128 g Sandisk ultra should be fin e't know what the issue was .
According to the manual you have to format the card in the recorder. You can't use formatted cards by windows or strait out of the package. Next there are memory cards and memory cards. You have to be keen on the fact if the card can handle the data stream, so you need cards special for this purpose. If not you could experience the effect you wrote about.
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Interesting B&H says new item - coming soon.
Wonder if that's just because of restocking...
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The Avisoft site has this device at number 8 (joint with the Portacapture devices) in their noise-ranked table. Above it, the Zoom F6, above that the F3. But the differences are neither here nor there.
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Interesting that the Fostex FR2LE, a 13 year old recorder, is number 3, above the Zooms.
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Interesting that the Fostex FR2LE, a 13 year old recorder, is number 3, above the Zooms.
It IS very quiet :clapping:
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Boy oh boy, I do feel embarresed. I got a response from Tascam regarding the file split issue...
The problem you describe could not be reconstructed. We have tested with several audio editors.
The FR-AV2 does not write any fade-outs or -ins, when splitting files. It will cut straight into the waveform, even when it is not at a zero point. All our test results were perfectly seemless cuts.
Therefore we believe that your editor app/DAW might automatically create short fade-ins, when encountering files that do not begin on zero point. Please check your app settings to change this preset.
Mit freundlichen Grüßen / Best Regards,
Stupid me... Now I checked the 2nd and 3rd file of a 'splitted recording' separately, so without appending them to the first. And indeed, all files don't have the fade-ins! It turns out it was caused by a setting in Audition called 'Smooth all edit boundaries by cross-fading x ms' (where x can be set by user, in my case it was set to 5ms). See attached picture. After disabling this option, doing an 'append' results in a seamless split! See other attached picture.
So for once and for all, and with my deepest apologies to Tascam and anybody reading this:
There are NO file split issues with the Tascam FR-AV2!
@Kyle: Did you manage to figure out your issues at the end of the files?
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Interesting. I am not quite sure what setting in REAPER would cause anything like I saw, I have not modified my workflow at all from when I was working with files with splits via A10 pretty regularly. My issue seemed different from yours though. Both my AV2s produced files with identical extreme clipping at the END of the break points, as opposed to the fades. Break points between devices were at different points in the show.
https://taperssection.com/index.php?topic=205834.msg2418945#msg2418945
The fact that they report clean splits is interesting. I will see if I can replicate at home but I'm not super optimistic -- as I said, I am pretty sure I didn't see this in home environment testing, only out in the field (SBD/Stage matrix)........ no idea why it would have occurred one place but not the other.
So yeah, I'll see if I can replicate with a closer eye & current settings, and if I can't, I guess I'll just have to pray at next shows!
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Interesting. Indeed.
Recently found the same in Reaper with a 192/24 file from a dr100mk3 - also not seamless and not having seen that before, but it wasn’t an issue in that case.
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Interesting. Indeed.
Recently found the same in Reaper with a 192/24 file from a dr100mk3 - also not seamless and not having seen that before, but it wasn’t an issue in that case.
I downloaded the latest REAPER 64bits for windows (7.27) and opened a couple of files created with the FR-AV2. All of them seem perfectly fine at the end.
Maybe it’s worth mentioning the version and platform where you get the problems? Or maybe share a problematic file to see how it behaves with other software?
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It has been a few months, if it occurs I'll start a new thread - now back to the FR-AV2 ;-)
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I was able to use mine last night for the first time when a friend called and offered me a ticket to City and Colour. Used the 3.5 mm input on 5V with CA11 cardioids. It wasn't an ideal taping location being way off to the side but I was within 25 feet of the stack. Turned out nice considering all I had was a SRS XY mount for the mics and how off center I was.
The deck performed as expected, mic power was adequate for a volume level that required earplugs pretty close to the stacks and battery life was far better than expected. I recorded both bands around 2h:45m and used less than half of the AAs (freshly charged but pretty old Eneloop 1900mah).
I'll get the show tracked and on my Soundcloud for sharing in the next couple days. Great show go see City and Colour if you can!
I bought this deck mainly for use with P48 full size mics but it performs quite well with CA11 on the ext in. No better or worse than the Korg MR1 or the Sony A10 I've been using but no battery box required so that's a plus. It's quite a bit larger than the A10 so if I had to be worried about getting it in I'd still go with the A10.
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Here's a Soundcloud link to the City and Colour show I recorded Sunday - https://on.soundcloud.com/AfuZKqXSHbPJaba39 (https://on.soundcloud.com/AfuZKqXSHbPJaba39) - no processing other than gain adjustments. I was off to the side about 20-25 feet away from the stack. CA11 cardioids in a SRS XY mount straight into the EXT IN with 5V PIP. On the last three songs I moved around a little - last two songs hugging the stack from about 10 feet away with the mic mount taped to the strap of my shoulder bag.
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I'd say yes, there's no analog gain.
For what it's worth--and maybe this has already been fully clarified elsewhere in this thread (which I admit I have not read from start to finish), in which case my apologies for the redundancy--there is no analog input stage gain control whatsoever on this new tascam unit. I emailed Tascam directly to ask about this (back in September), since the manual's section on 'Setting the Recording Level' was not clear, and their answer was "REC LEVEL is done at the digital stage"
which, to my mind, makes it unnecessary to try to adjust the REC LEVEL while recording, as there is no advantage or benefit to doing so then vs. later, in post.
If anyone else has more current info from Tascam on analog gain control / REC LEVEL domain that contradicts the answer I received from them, please let me/us know!
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It seems to me that having analog input level on a 32 bit float recorder does risk undermining the whole point. I do feel that a key benefit of 32 bit float devices is that the whole signal chain can be optimised end to end - the mic (if built in) can be matched to the preamp, and the preamp matched to the a/d converter, using the maximum preamp gain which the a/d stage can handle, because max preamp gain gives lowest noise. On the very modest 32 bit float devices I have, the chosen level seems to be what I would have chosen myself, if that makes sense. It's almost as if the device is using auto gain control, but fixed, and set by mind reading and time travel to the correct setting at the outset...
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I'd say yes, there's no analog gain.
For what it's worth--and maybe this has already been fully clarified elsewhere in this thread (which I admit I have not read from start to finish), in which case my apologies for the redundancy--there is no analog input stage gain control whatsoever on this new tascam unit. I emailed Tascam directly to ask about this (back in September), since the manual's section on 'Setting the Recording Level' was not clear, and their answer was "REC LEVEL is done at the digital stage"
which, to my mind, makes it unnecessary to try to adjust the REC LEVEL while recording, as there is no advantage or benefit to doing so then vs. later, in post.
If anyone else has more current info from Tascam on analog gain control / REC LEVEL domain that contradicts the answer I received from them, please let me/us know!
Zoom makes this a bit clearer in the F6 and F8N PRO manuals. Those recorders can do both 24-bit fixed and 32-bit float record modes. In either mode, the analog input gain is fixed. All level adjustments (trim, fader, etc) happen post-ADC in the digital domain. The mode just determines when that level adjustment happens, the signal routing, and file format.
I suspect the FR-AV2 and other 32FP-capable units from Tascam operate a similar way.
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Let it run with eneloop pros again, 24v phantom power via XLRs powering DPA 4060s - ran 9 hours and the file splits were clean. Cannot explain the distortion I saw at the end of the files from when I was actually taping a show. Did not modify any settings or update any firmware since. Same DAW flow.
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I'd say yes, there's no analog gain.
For what it's worth--and maybe this has already been fully clarified elsewhere in this thread (which I admit I have not read from start to finish), in which case my apologies for the redundancy--there is no analog input stage gain control whatsoever on this new tascam unit. I emailed Tascam directly to ask about this (back in September), since the manual's section on 'Setting the Recording Level' was not clear, and their answer was "REC LEVEL is done at the digital stage"
which, to my mind, makes it unnecessary to try to adjust the REC LEVEL while recording, as there is no advantage or benefit to doing so then vs. later, in post.
If anyone else has more current info from Tascam on analog gain control / REC LEVEL domain that contradicts the answer I received from them, please let me/us know!
Thanks for forwarding the info provided by Tascam! Just a small addition: Although there is no input stage gain control, there are of course three different input paths into this recorder (MIC, LINE and the 3.5mm EXT input), each with their own fixed input stage gain. It makes me wonder if one of them is fed directly into the 2xADC, and if the other two are amplified/attenuated to get the same dynamic range before feeding them to the 2xADC...
On this matter, I am puzzled why the input ratings are defined differently in the manual for MIC, LINE and EXT.
For MIC, they mention a max and min level in dBu (no nominal).
For LINE, they mention a max and nominal level in dBu (no min).
For EXT they mention a max, nominal and min in dBV.
Very confusing for this digitally-oriented guy... Can someone maybe explain how to read this? I think I understand defining a max level, but why min and nominal?? What do these mean, why do they matter?
(If I'm correct, dBu is typically used for professional equipment and dBV for consumer equipement, and one should add 2.21 to dBV to convert to dBu, right?)
When MIC input selected
- Maximum input level: +4 dBu
- Minimum input level: −76 dBu
- Input impedance: 2.0 kΩ or more
When LINE input selected
- Maximum input level: +24 dBu
- Nominal input level: +4 dBu (GAIN setting at minimum)
- Input impedance: 8 kΩ or more
EXT
- Maximum input level: 1 dBV
- Nominal input level: −19 dBV (GAIN setting at minimum)
- Minimum input level: −79 dBV
- Input impedance: 6 kΩ or higher (when plug-in power is off )
1.6 kΩ or higher (when plug-in power is on)
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I emailed Tascam directly to ask about this (back in September), since the manual's section on 'Setting the Recording Level' was not clear, and their answer was "REC LEVEL is done at the digital stage"
If anyone else has more current info from Tascam on analog gain control / REC LEVEL domain that contradicts the answer I received from them, please let me/us know!
On this matter, I am puzzled why the input ratings are defined differently in the manual for MIC, LINE and EXT.
For MIC, they mention a max and min level in dBu (no nominal).
For LINE, they mention a max and nominal level in dBu (no min).
For EXT they mention a max, nominal and min in dBV.
Very confusing for this digitally-oriented guy... Can someone maybe explain how to read this? I think I understand defining a max level, but why min and nominal?? What do these mean, why do they matter?
(If I'm correct, dBu is typically used for professional equipment and dBV for consumer equipement, and one should add 2.21 to dBV to convert to dBu, right?)
When MIC input selected
- Maximum input level: +4 dBu
- Minimum input level: −76 dBu
- Input impedance: 2.0 kΩ or more
When LINE input selected
- Maximum input level: +24 dBu
- Nominal input level: +4 dBu (GAIN setting at minimum)
- Input impedance: 8 kΩ or more
EXT
- Maximum input level: 1 dBV
- Nominal input level: −19 dBV (GAIN setting at minimum)
- Minimum input level: −79 dBV
- Input impedance: 6 kΩ or higher (when plug-in power is off )
1.6 kΩ or higher (when plug-in power is on)
There are two things here, one is that Tascam already use "rec level in digital stage" for a longer time, but a different digital stage then most think. As an example, on the DR-701D they use probably encoders as volume potmeters for each channel. That's clever, you kick out a lot of problems seen with analog potmeters. The input is going probably through an IC with part number like NJW1195(A). This is a 4-Channel Electronic Volume with Input Selector. In this way you can have different amplifier setups, different gains and input impedance's for different sources.
With MIC they use a minimum specification, because its more related to the specs of a microphone, you can see directly if you can use that microphone or not.
With LINE they use nominal because +4 dBu is widely used in studio and other "pro"-settings. A specification of a minimum is not that useful for a line input.
With EXT the nominal level is about the standard of DIN, the maximum is a little less than the maximum for MIC. This amplifier seems to have a gain of 20 dB. The minimum is the lowest signal which can be handled, with 20 dB gain you get then -59 dBV as maximum. It looks like the EXT input also switches between a mic or a line setting when pip is used.
I hope this will clear a few things.
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I emailed Tascam directly to ask about this (back in September), since the manual's section on 'Setting the Recording Level' was not clear, and their answer was "REC LEVEL is done at the digital stage"
If anyone else has more current info from Tascam on analog gain control / REC LEVEL domain that contradicts the answer I received from them, please let me/us know!
On this matter, I am puzzled why the input ratings are defined differently in the manual for MIC, LINE and EXT.
For MIC, they mention a max and min level in dBu (no nominal).
For LINE, they mention a max and nominal level in dBu (no min).
For EXT they mention a max, nominal and min in dBV.
Very confusing for this digitally-oriented guy... Can someone maybe explain how to read this? I think I understand defining a max level, but why min and nominal?? What do these mean, why do they matter?
(If I'm correct, dBu is typically used for professional equipment and dBV for consumer equipement, and one should add 2.21 to dBV to convert to dBu, right?)
When MIC input selected
- Maximum input level: +4 dBu
- Minimum input level: −76 dBu
- Input impedance: 2.0 kΩ or more
When LINE input selected
- Maximum input level: +24 dBu
- Nominal input level: +4 dBu (GAIN setting at minimum)
- Input impedance: 8 kΩ or more
EXT
- Maximum input level: 1 dBV
- Nominal input level: −19 dBV (GAIN setting at minimum)
- Minimum input level: −79 dBV
- Input impedance: 6 kΩ or higher (when plug-in power is off )
1.6 kΩ or higher (when plug-in power is on)
There are two things here, one is that Tascam already use "rec level in digital stage" for a longer time, but a different digital stage then most think. As an example, on the DR-701D they use probably encoders as volume potmeters for each channel. That's clever, you kick out a lot of problems seen with analog potmeters. The input is going probably through an IC with part number like NJW1195(A). This is a 4-Channel Electronic Volume with Input Selector. In this way you can have different amplifier setups, different gains and input impedance's for different sources.
With MIC they use a minimum specification, because its more related to the specs of a microphone, you can see directly if you can use that microphone or not.
With LINE they use nominal because +4 dBu is widely used in studio and other "pro"-settings. A specification of a minimum is not that useful for a line input.
With EXT the nominal level is about the standard of DIN, the maximum is a little less than the maximum for MIC. This amplifier seems to have a gain of 20 dB. The minimum is the lowest signal which can be handled, with 20 dB gain you get then -59 dBV as maximum. It looks like the EXT input also switches between a mic or a line setting when pip is used.
I hope this will clear a few things.
That’s great information. Thank you!
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That’s great information. Thank you!
Thanks, I don't know what they did within the FR-AV2. Probably something very clever as usual.
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There are two things here, one is that Tascam already use "rec level in digital stage" for a longer time, but a different digital stage then most think. As an example, on the DR-701D they use probably encoders as volume potmeters for each channel. That's clever, you kick out a lot of problems seen with analog potmeters. The input is going probably through an IC with part number like NJW1195(A). This is a 4-Channel Electronic Volume with Input Selector. In this way you can have different amplifier setups, different gains and input impedance's for different sources.
With MIC they use a minimum specification, because its more related to the specs of a microphone, you can see directly if you can use that microphone or not.
With LINE they use nominal because +4 dBu is widely used in studio and other "pro"-settings. A specification of a minimum is not that useful for a line input.
With EXT the nominal level is about the standard of DIN, the maximum is a little less than the maximum for MIC. This amplifier seems to have a gain of 20 dB. The minimum is the lowest signal which can be handled, with 20 dB gain you get then -59 dBV as maximum. It looks like the EXT input also switches between a mic or a line setting when pip is used.
I hope this will clear a few things.
Thanks Joop for your brave attempt to educate me! I've been reading your response a couple of times and I think I largely understand :)
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I opened a support ticket regarding being unable to turn the screen off while recording with Phantom Power.
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I'd say yes, there's no analog gain.
For what it's worth--and maybe this has already been fully clarified elsewhere in this thread (which I admit I have not read from start to finish), in which case my apologies for the redundancy--there is no analog input stage gain control whatsoever on this new tascam unit. I emailed Tascam directly to ask about this (back in September), since the manual's section on 'Setting the Recording Level' was not clear, and their answer was "REC LEVEL is done at the digital stage"
which, to my mind, makes it unnecessary to try to adjust the REC LEVEL while recording, as there is no advantage or benefit to doing so then vs. later, in post.
If anyone else has more current info from Tascam on analog gain control / REC LEVEL domain that contradicts the answer I received from them, please let me/us know!
Zoom makes this a bit clearer in the F6 and F8N PRO manuals. Those recorders can do both 24-bit fixed and 32-bit float record modes. In either mode, the analog input gain is fixed. All level adjustments (trim, fader, etc) happen post-ADC in the digital domain. The mode just determines when that level adjustment happens, the signal routing, and file format.
I suspect the FR-AV2 and other 32FP-capable units from Tascam operate a similar way.
Hi Voltronic--
I also took the time to recently ask Zoom about where the trim (gain) control operates in both the F6 and F8n Pro, when operating in linear mode (which is the only mode I use).
(This because I had stumbled across your explanation of the F6's signal flow (above) on another platform, which was I believe based on one of the block diagrams in the F6 manual. From there I looked at the manual and and block diagrams for the F8n Pro, which led me to looking at the manuals for the F8 and F8n as well, and comparing them all, where I realized that there were differences in the block diagrams across the 4 resources.)
Zoom's response was that, for both the F6 and F8n Pro in linear mode, the signal flow was XLR->Trim->ADC. That answer matches most closely to the Detailed Product Diagram on p130 of the F8 manual and p185 of the F8n manual. I didn't question them further as to why the the F8n Pro and F6 manuals contains dissimilar graphics / don't spell this out clearly.
In addition, they also clarified that the dual ADC system operates in both linear and 32bf mode.
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I opened a support ticket regarding being unable to turn the screen off while recording with Phantom Power.
I thought that was just me not knowing what I had done in the menu on my first outing with it.
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I opened a support ticket regarding being unable to turn the screen off while recording with Phantom Power.
I'm still in doubt about opening a ticket about not being able to turn down/off the screen when powered through USB, especially after my rather embaressing faulty ticket about the track splitting...
It works as described in the manual (all user display settings get overruled when powering through USB), but I still think this is a really weird and annoying 'feature' that I'd like to get removed.
The urgency for my 'request' has lowered quite a bit since I know the battery life is rather good and sufficient for my personal purposes. I don't think I will ever need to run the device on a power bank.
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Just a passing thought on the need to power these devices for long periods - given that in most cases one can now simply turn on the rig and walk away, it does enable live recordings to be made by setting up the gear before the public are admitted (if possible), setting it running, and collecting it at the end of the night. So external powering might be more necessary than in other scenarios. In fact, you might be more worried about losing power than clipping the recording. Of course there is always the issue of someone helping themselves to the gear when unattended, I admit.
I once recorded three classical concerts at the same time in London, in three different halls, though admittedly two were in the same complex. That absolutely required the set-and-forget approach, and it was long before 32 bit float was even thought of! But I got away with it...
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Used my FR AV2 for a taping outing with P48 mics for the first time - https://on.soundcloud.com/VLJYCuPhFS6vaGqU7 (https://on.soundcloud.com/VLJYCuPhFS6vaGqU7)
King Gizzard 2024-11-12 Criterion, Oklahoma City, OK
MBHO KA500 hypers with Naiant PFA plugged straight in. Mics on an extension pole clamped to the front left corner of the SBD enclosure (so I was off center 20 feet or more and you can tell) about 7 ft high.
I used some EQ on the bass - rolling off quite a bit as I always do - the low extension of the preamp in this recorder is quite good.
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Which Input(s) would you prefer using when connecting a soundboard to the FR-AV2 ?
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The XLR/TRS (LINE setting) have the highest max input and are balanced, so I opt for those in terms of minmaxing.
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The XLR/TRS (LINE setting) have the highest max input and are balanced, so I opt for those in terms of minmaxing.
Totally agree, if the board operator is letting you have an XLR send.
Some boards like X32 have RCA or 1/4" outputs available.
If the 1/4" are balanced (three conductor, "TRS" = Tip-Ring-Sleeve) then it's beneficial to use TRS > XLR cabling and keep the connection balanced.
Otherwise, for unbalanced 1/4" or RCA outputs, any cable with proper connectors will be sufficient, and it may be most convenient to use 1/8" line inputs if you don't mind them not locking like XLR's.
With a board patch, it's not like it's gonna get bumped around in your bag, so the locking connector should be less of a worry. Plus you might already have that cable and not have a pair of TRS > XLR handy.
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The XLR/TRS (LINE setting) have the highest max input and are balanced, so I opt for those in terms of minmaxing.
Totally agree, if the board operator is letting you have an XLR send.
Some boards like X32 have RCA or 1/4" outputs available.
If the 1/4" are balanced (three conductor, "TRS" = Tip-Ring-Sleeve) then it's beneficial to use TRS > XLR cabling and keep the connection balanced.
Otherwise, for unbalanced 1/4" or RCA outputs, any cable with proper connectors will be sufficient, and it may be most convenient to use 1/8" line inputs if you don't mind them not locking like XLR's.
With a board patch, it's not like it's gonna get bumped around in your bag, so the locking connector should be less of a worry. Plus you might already have that cable and not have a pair of TRS > XLR handy.
If the output from the board is 1/4" balanced then it would be best to go TRS>TRS so it goes into the 1/4" line level input of the recorder. Line in on that unit is only 1/4" or unbalanced 1/8" stereo mini. The XLR inputs are microphone level only.
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The XLR/TRS (LINE setting) have the highest max input and are balanced, so I opt for those in terms of minmaxing.
Totally agree, if the board operator is letting you have an XLR send.
Some boards like X32 have RCA or 1/4" outputs available.
If the 1/4" are balanced (three conductor, "TRS" = Tip-Ring-Sleeve) then it's beneficial to use TRS > XLR cabling and keep the connection balanced.
Otherwise, for unbalanced 1/4" or RCA outputs, any cable with proper connectors will be sufficient, and it may be most convenient to use 1/8" line inputs if you don't mind them not locking like XLR's.
With a board patch, it's not like it's gonna get bumped around in your bag, so the locking connector should be less of a worry. Plus you might already have that cable and not have a pair of TRS > XLR handy.
If the output from the board is 1/4" balanced then it would be best to go TRS>TRS so it goes into the 1/4" line level input of the recorder. Line in on that unit is only 1/4" or unbalanced 1/8" stereo mini. The XLR inputs are microphone level only.
Thanks for your suggestions ! :).
How do i connect a Stereo 1/4" output from the soundboard to the Tascam to the 1/4" connector(s) ?
1: Do i have to use BOTH TRS Inputs of the Tascam ( i would need a y bracket cable that seperates Left and Right Channel from the soundboard output - is this balanced anymore? )?
2: Or is one of TRS Ports of the Tascam capable of recording a Stereo Signal (so i could just use an ordinary 1xTRS to 1x TRS cable)?
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The XLR/TRS (LINE setting) have the highest max input and are balanced, so I opt for those in terms of minmaxing.
Totally agree, if the board operator is letting you have an XLR send.
Some boards like X32 have RCA or 1/4" outputs available.
If the 1/4" are balanced (three conductor, "TRS" = Tip-Ring-Sleeve) then it's beneficial to use TRS > XLR cabling and keep the connection balanced.
Otherwise, for unbalanced 1/4" or RCA outputs, any cable with proper connectors will be sufficient, and it may be most convenient to use 1/8" line inputs if you don't mind them not locking like XLR's.
With a board patch, it's not like it's gonna get bumped around in your bag, so the locking connector should be less of a worry. Plus you might already have that cable and not have a pair of TRS > XLR handy.
If the output from the board is 1/4" balanced then it would be best to go TRS>TRS so it goes into the 1/4" line level input of the recorder. Line in on that unit is only 1/4" or unbalanced 1/8" stereo mini. The XLR inputs are microphone level only.
Thanks for your suggestions ! :).
How do i connect a Stereo 1/4" output from the soundboard to the Tascam to the 1/4" connector(s) ?
1: Do i have to use BOTH TRS Inputs of the Tascam ( i would need a y bracket cable that seperates Left and Right Channel from the soundboard output - is this balanced anymore? )?
2: Or is one of TRS Ports of the Tascam capable of recording a Stereo Signal (so i could just use an ordinary 1xTRS to 1x TRS cable)?
For consumer electronics the mindset is that TRS means stereo. Not so in pro/PA gear*. TRS means balanced, just like an XLR cable. The TRS 1/4" out of the soundboard is not going to be a single jack carrying both L/R, there will be two of them, one for left and one for right. So you'd use two TRS 1/4" cables, one for each input on the Tascam.
*I mean obviously a 1/4" headphone jack/plug is TRS and that is used in pro gear.
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Thanks!
One more question:
On the ZOOM F3 theres no gain settings for 32bit float recordings anymore (can/should be regulated in the post anyhow) and i was very happy that clipping and setting the gain during stealth-recording is not a problem anymore.
With the Tascam you are forced to set a gain on the input level of the mics again - why is that so? ???
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In 32 bit float you can input a "gain" as such on both recorders but it is pretty inconsequential if you go under or over.
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On the ZOOM F3 theres no gain settings for 32bit float recordings anymore (can/should be regulated in the post anyhow) and i was very happy that clipping and setting the gain during stealth-recording is not a problem anymore.
With the Tascam you are forced to set a gain on the input level of the mics again - why is that so? ???
Input amplifiers don't have unlimited input voltage. With Zoom F3 its +4dBu, after that you have a problem, 32-bit float or not. With Tascam you have to set the record level to "normal" levels, in case of a very loud noise you have room to process that without clipping. Not the same as with Zoom, there you can clip. The room in 32-bit float is not usable because there are no "op amps" which can process that kind of dynamic range, same for the AD converter. At best the dynamic room will be somewhere between 130 dB to a little more then 140 dB, depending on what ADC is used. Above that you will have distortion, no matter what.
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Ok, its a safety-net to level down the analogue input stages to prevent them from clipping on very loud shows and maybe somewhat limited mics.
So the factory preset is +4dB - fixed on the zoom vs. +16dB - changeable on the tascam.
i never had an issue with clipping using the zoom f3 so far.
i guess setting a more conservative +4dB on the tascam as well is the safer setting or can this lead to more noise if i have to level up the recording in case its very quiet?
thanks for your input/help :)
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So the factory preset is +4dB - fixed on the zoom vs. +16dB - changeable on the tascam.
i guess setting a more conservative +4dB on the tascam as well is the safer setting or can this lead to more noise if i have to level up the recording in case its very quiet?
According Curtis Judd the Zoom F3 has 10dB more noise compared to the Tascam FR-AV2. I don't know where you get that +16dB from, the maximum input for mic with the Tascam is also +4dBu. For setting level on meters, set the nominal level at -12dB or try -6dB when recording in 32-bit float mode. You won't hear noise from the Tascam, so if you hear noise its time to investigate in a better set of mics or if on TRS go to XLR. Cranking up the record volume does not change the noise, only amplify it.
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Thanks!
One more question:
On the ZOOM F3 theres no gain settings for 32bit float recordings anymore (can/should be regulated in the post anyhow) and i was very happy that clipping and setting the gain during stealth-recording is not a problem anymore.
With the Tascam you are forced to set a gain on the input level of the mics again - why is that so? ???
I’m not sure if the previous responses really answered your question clearly, so to avoid any confusion:
You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
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If the output from the board is 1/4" balanced then it would be best to go TRS>TRS so it goes into the 1/4" line level input of the recorder. Line in on that unit is only 1/4" or unbalanced 1/8" stereo mini. The XLR inputs are microphone level only.
haha you caught me not owning the unit or reading the manual! Thanks for the correction! Makes a big difference, I'm confident!
Yeah, use 1/4 TRS to 1/4 TRS if possible.
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
It’s good to have discussion, and I must admit I assumed recording at 32-bit floating point. However, even when recording at 24 bits it doesn’t necessarily mean there must be analog gain control. The device can operate completely identical regardless recording to 24 bit fixed point or 32 bit floating point storage, apart from that in 24-bit the samples for the output file are converted from the internally used 32bfp to 24bit before storing. And in that case, it would be wise to make sure (by setting the fully digital ‘record level’) that the samples are not too quiet and won’t go over 0dbFS.
It seems Tascam has confirmed there is no analog gain control whatsoever on this device, regardless of the used storage format. See https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100 (https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100).
Of course the dual-ADC setup helps to properly deal with both quiet and loud signals.
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The XLR/TRS (LINE setting) have the highest max input and are balanced, so I opt for those in terms of minmaxing.
Totally agree, if the board operator is letting you have an XLR send.
Some boards like X32 have RCA or 1/4" outputs available.
If the 1/4" are balanced (three conductor, "TRS" = Tip-Ring-Sleeve) then it's beneficial to use TRS > XLR cabling and keep the connection balanced.
Otherwise, for unbalanced 1/4" or RCA outputs, any cable with proper connectors will be sufficient, and it may be most convenient to use 1/8" line inputs if you don't mind them not locking like XLR's.
With a board patch, it's not like it's gonna get bumped around in your bag, so the locking connector should be less of a worry. Plus you might already have that cable and not have a pair of TRS > XLR handy.
If the output from the board is 1/4" balanced then it would be best to go TRS>TRS so it goes into the 1/4" line level input of the recorder. Line in on that unit is only 1/4" or unbalanced 1/8" stereo mini. The XLR inputs are microphone level only.
Huh? Where is this coming from? You are able to set it to LINE level and record via XLR - max input is +24 dBu regardless of XLR vs TRS when set to LINE, no?
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According Curtis Judd the Zoom F3 has 10dB more noise compared to the Tascam FR-AV2.
I don't know where he gets that from - according to the very respected Avisoft measurements Equivalent Input Noise (EIN)
impedance: 150 ohms for the two devices is within 1dBu with the F3 being that tiny bit in front.
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
It’s good to have discussion, and I must admit I assumed recording at 32-bit floating point. However, even when recording at 24 bits it doesn’t necessarily mean there must be analog gain control. The device can operate completely identical regardless recording to 24 bit fixed point or 32 bit floating point storage, apart from that in 24-bit the samples for the output file are converted from the internally used 32bfp to 24bit before storing. And in that case, it would be wise to make sure (by setting the fully digital ‘record level’) that the samples are not too quiet and won’t go over 0dbFS.
It seems Tascam has confirmed there is no analog gain control whatsoever on this device, regardless of the used storage format. See https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100 (https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100).
Of course the dual-ADC setup helps to properly deal with both quiet and loud signals.
A portion from the manual is attached here that kind of confirms too that there is no analog gain control. If there is analog overload, they advice to lower the volume of the source or to move the mics away from the source. They do not tell you to lower the 'record level', as that won't help to relieve the analog part.
It's a pitty you cannot see how much you're away from analog overload while recording, only when you've hit the limit... :(
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The XLR/TRS (LINE setting) have the highest max input and are balanced, so I opt for those in terms of minmaxing.
Totally agree, if the board operator is letting you have an XLR send.
Some boards like X32 have RCA or 1/4" outputs available.
If the 1/4" are balanced (three conductor, "TRS" = Tip-Ring-Sleeve) then it's beneficial to use TRS > XLR cabling and keep the connection balanced.
Otherwise, for unbalanced 1/4" or RCA outputs, any cable with proper connectors will be sufficient, and it may be most convenient to use 1/8" line inputs if you don't mind them not locking like XLR's.
With a board patch, it's not like it's gonna get bumped around in your bag, so the locking connector should be less of a worry. Plus you might already have that cable and not have a pair of TRS > XLR handy.
If the output from the board is 1/4" balanced then it would be best to go TRS>TRS so it goes into the 1/4" line level input of the recorder. Line in on that unit is only 1/4" or unbalanced 1/8" stereo mini. The XLR inputs are microphone level only.
Huh? Where is this coming from? You are able to set it to LINE level and record via XLR - max input is +24 dBu regardless of XLR vs TRS when set to LINE, no?
I thought mepaca was right about that. I too thought that the FR-AV2 had three completely independent pairs of inputs TRS/LINE, XLR/MIC and minijack/EXT, each with their own electrical characteristics, and you had to choose which of these three you want to use during recording through the setup menu. However, I just tested this by connecting my bass guitar (the only thing I have at home with TRS-ish connector) to TRS, and then switch the input between LINE and MIC. Lo-and-behold, with both I see the meters go when playing the strings! So it seems it works just like the Zoom F3 where you can plug into XLR or TRS and select LINE/MIC from the input setup menu.
Edit: originally I made an incorrect remark here about the Zoom F3 having TRS inputs, which is not the case. I must have mixed up the H1XLR… Thanks for pointing this out!
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One difference though: On the Tascam you cannot enable phantom power on the TRS plugs (only on XLR), while with the Zoom F3 you can...
No TRS on the Zoom F3 AFAIK.
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Concur with the above post from Rairun
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One difference though: On the Tascam you cannot enable phantom power on the TRS plugs (only on XLR), while with the Zoom F3 you can...
No TRS on the Zoom F3 AFAIK.
Sorry, my bad. Been reading too many spec sheets and manuals lately. I shouldn’t say things about devices i don’t own…
Will amend…
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New firmware is out! No really spectacular changes…
Firmware V1.03
The frame rate can now be changed at any time, except when the timecode is being synchronized from an external source.The menu structure and operation have also been updated accordingly. For details on the changes, please refer to "15. Timecode functions" in the Owner's Manual.
Operation stability has been improved in other ways.
See https://tascam.com/us/product/fr-av2/support (https://tascam.com/us/product/fr-av2/support)
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
It’s good to have discussion, and I must admit I assumed recording at 32-bit floating point. However, even when recording at 24 bits it doesn’t necessarily mean there must be analog gain control. The device can operate completely identical regardless recording to 24 bit fixed point or 32 bit floating point storage, apart from that in 24-bit the samples for the output file are converted from the internally used 32bfp to 24bit before storing. And in that case, it would be wise to make sure (by setting the fully digital ‘record level’) that the samples are not too quiet and won’t go over 0dbFS.
It seems Tascam has confirmed there is no analog gain control whatsoever on this device, regardless of the used storage format. See https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100 (https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100).
Of course the dual-ADC setup helps to properly deal with both quiet and loud signals.
A portion from the manual is attached here that kind of confirms too that there is no analog gain control. If there is analog overload, they advice to lower the volume of the source or to move the mics away from the source. They do not tell you to lower the 'record level', as that won't help to relieve the analog part.
Indeed, it was these proposed solutions from this page in the manual that led me to seek clarification from Tascam on whether there was, or was not, gain control in the analog realm. I suppose there is a 3rd option available for those whose mics (like my c480b's) have output pads: enable the pads.
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I bought it on a whim and have way too much gear already.
Possibly you are not the first person here to experience those sentiments, regardless of what the latest purchase was...!
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
It’s good to have discussion, and I must admit I assumed recording at 32-bit floating point. However, even when recording at 24 bits it doesn’t necessarily mean there must be analog gain control. The device can operate completely identical regardless recording to 24 bit fixed point or 32 bit floating point storage, apart from that in 24-bit the samples for the output file are converted from the internally used 32bfp to 24bit before storing. And in that case, it would be wise to make sure (by setting the fully digital ‘record level’) that the samples are not too quiet and won’t go over 0dbFS.
It seems Tascam has confirmed there is no analog gain control whatsoever on this device, regardless of the used storage format. See https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100 (https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100).
Of course the dual-ADC setup helps to properly deal with both quiet and loud signals.
A portion from the manual is attached here that kind of confirms too that there is no analog gain control. If there is analog overload, they advice to lower the volume of the source or to move the mics away from the source. They do not tell you to lower the 'record level', as that won't help to relieve the analog part.
Indeed, it was these proposed solutions from this page in the manual that led me to seek clarification from Tascam on whether there was, or was not, gain control in the analog realm. I suppose there is a 3rd option available for those whose mics (like my c480b's) have output pads: enable the pads.
Wouldn't switching to 'line input' practically do the same thing? Of course that would only work for the TRS or XLR inputs, not for the 3.5mm EXT input.
I have been wondering about finding out 'how far we'd be from analog overload' during recording, e.g. when using sensitive mics for loud rock shows. I was thinking that if you'd set the 'record level' to +0dB, you might get analog overload if you then go over 0dB on your meters. I did a test and it seems I was right about this: I set the 'record level' to +0dB, connected a pair of Clippy mics to the recorder and put the mics in front of my home stereo speakers. Then I slowly cranked up the volume and once the meters hit the 0dB on the level indicators, the complete VU bars turned red, meaning analog overload according to the manual. Cool!
You don't necessarily need to set the record level to +0dB to make use of this knowledge. E.g. if we set the level to +20dB, we now know we won't get analog overload as long as the VU bars stay below +20dB. (I know, they only go to +6dB, but just to paint the picture...)
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I bought it on a whim and have way too much gear already.
Possibly you are not the first person here to experience those sentiments, regardless of what the latest purchase was...!
I thought it would be a good solution to having a small recorder to run my mini mics without a battery box but honestly for the few occasions I need to be small the A10 and battery box does just fine and I have a DR100 with an Oade preamp mod or a Mixpre6 to use for P48 mics. I need to shift some of my gear $ into my multitrack rig - I had to buy a snake and an ADAT interface to get my channel count up to 18 tracks.
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
It’s good to have discussion, and I must admit I assumed recording at 32-bit floating point. However, even when recording at 24 bits it doesn’t necessarily mean there must be analog gain control. The device can operate completely identical regardless recording to 24 bit fixed point or 32 bit floating point storage, apart from that in 24-bit the samples for the output file are converted from the internally used 32bfp to 24bit before storing. And in that case, it would be wise to make sure (by setting the fully digital ‘record level’) that the samples are not too quiet and won’t go over 0dbFS.
It seems Tascam has confirmed there is no analog gain control whatsoever on this device, regardless of the used storage format. See https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100 (https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100).
Of course the dual-ADC setup helps to properly deal with both quiet and loud signals.
A portion from the manual is attached here that kind of confirms too that there is no analog gain control. If there is analog overload, they advice to lower the volume of the source or to move the mics away from the source. They do not tell you to lower the 'record level', as that won't help to relieve the analog part.
Indeed, it was these proposed solutions from this page in the manual that led me to seek clarification from Tascam on whether there was, or was not, gain control in the analog realm. I suppose there is a 3rd option available for those whose mics (like my c480b's) have output pads: enable the pads.
Wouldn't switching to 'line input' practically do the same thing? Of course that would only work for the TRS or XLR inputs, not for the 3.5mm EXT input.
Unless I am not recalling it correctly: unlike the F3, +48v power is not available for 'Line In' mode on the AV2. This difference is where the F3 has an advantage over the Tascam unit, when it comes to using mics that require phantom power.
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
It’s good to have discussion, and I must admit I assumed recording at 32-bit floating point. However, even when recording at 24 bits it doesn’t necessarily mean there must be analog gain control. The device can operate completely identical regardless recording to 24 bit fixed point or 32 bit floating point storage, apart from that in 24-bit the samples for the output file are converted from the internally used 32bfp to 24bit before storing. And in that case, it would be wise to make sure (by setting the fully digital ‘record level’) that the samples are not too quiet and won’t go over 0dbFS.
It seems Tascam has confirmed there is no analog gain control whatsoever on this device, regardless of the used storage format. See https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100 (https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100).
Of course the dual-ADC setup helps to properly deal with both quiet and loud signals.
A portion from the manual is attached here that kind of confirms too that there is no analog gain control. If there is analog overload, they advice to lower the volume of the source or to move the mics away from the source. They do not tell you to lower the 'record level', as that won't help to relieve the analog part.
Indeed, it was these proposed solutions from this page in the manual that led me to seek clarification from Tascam on whether there was, or was not, gain control in the analog realm. I suppose there is a 3rd option available for those whose mics (like my c480b's) have output pads: enable the pads.
Wouldn't switching to 'line input' practically do the same thing? Of course that would only work for the TRS or XLR inputs, not for the 3.5mm EXT input.
Unless I am not recalling it correctly: unlike the F3, +48v power is not available for 'Line In' mode on the AV2. This difference is where the F3 has an advantage over the Tascam unit, when it comes to using mics that require phantom power.
And correct me if I'm wrong, but the Zoom F3 doesn't even have a 'Line In' mode as such, which would skip the preamp. It's essentially the same analogue path but with a -20 dB pad in front of it. It serves the same purpose when managing levels (i.e. the use case we are discussing here), but the device is not really made for people who for example want to use an external preamp for colouration or less noise.
But then again, I'd think the AV2 would be the same? Line in mode should just be a pad too? Which makes the absence of phantom power a little baffling.
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The F3's preamps are pretty darn quiet, so I certainly don't feel the need to put an additional external preamp in front of that device.
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people use external preamps because they like how they color the sound.
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people use external preamps because they like how they color the sound.
While this might be true some of the time, there are lots of people who use external preamps just as often because they like how they don't color the sound and/or provide other benefits. The external preamp might have higher gain, lower noise, more extended and flat frequency response, or a host of other things that are superior to the built-in preamps, which are often compromised out of necessity to fit a price point and form factor. There are some expensive preamps from brands like Neve or API that may purposely add color, but then you have brands like Pueblo, Millenia, and Grace which are known for ultimate transparency.
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Totally agree with Voltranic. Me personally, I would prefer transparency. Of course, YMMV. To each his or her own.
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It seems to me that in recent years, the chances of just about any listener being able to spot differences in noise or frequency response between recent recorders of good repute, and external preamps, are very low. In the context of recording live shows from audience positions, very very low. The mics used, and their placement, will be a hundred times more noticeable.
It seems to me... but maybe it does not seem to anyone else!
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are the preamps in the FR-AV2 the same as the Tascam DR-70D?
They both have what they refer to as their" HDDA preamp", but on the Tascam website, they refer to the preamps in the FR-AV2 as "Ultra HDDA" instead of just HDDA, like what is listed for the DR-70D.
There is a picture of the chip with the DR-70D board on the Tascam website: https://tascam.com/us/product/dr-70d
But I wasn't able to find the picture listed for the FR-AV2 : https://tascam.com/us/product/fr-av2
any thoughts on this?
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are the preamps in the FR-AV2 the same as the Tascam DR-70D?
They both have what they refer to as their" HDDA preamp", but on the Tascam website, they refer to the preamps in the FR-AV2 as "Ultra HDDA" instead of just HDDA, like what is listed for the DR-70D.
There is a picture of the chip with the DR-70D board on the Tascam website: https://tascam.com/us/product/dr-70d
But I wasn't able to find the picture listed for the FR-AV2 : https://tascam.com/us/product/fr-av2
any thoughts on this?
The HDDA preamp was used in the DR701d but I thought was an upgrade from the 70d. I could be, and likely am, wrong about the 70d. I had a 701 but not a 70.
EDIT- I followed the links and it looks like they upgraded the 70 with the mkii version (red handles) and included the HDDA preamp circuit.
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are the preamps in the FR-AV2 the same as the Tascam DR-70D?
They both have what they refer to as their" HDDA preamp", but on the Tascam website, they refer to the preamps in the FR-AV2 as "Ultra HDDA" instead of just HDDA, like what is listed for the DR-70D.
There is a picture of the chip with the DR-70D board on the Tascam website: https://tascam.com/us/product/dr-70d
But I wasn't able to find the picture listed for the FR-AV2 : https://tascam.com/us/product/fr-av2
any thoughts on this?
The HDDA preamp was used in the DR701d but I thought was an upgrade from the 70d. I could be, and likely am, wrong about the 70d. I had a 701 but not a 70.
EDIT- I followed the links and it looks like they upgraded the 70 with the mkii version (red handles) and included the HDDA preamp circuit.
Tascam products page identifying preamp type by product can be found here: https://tascam.com/us/contents/mic_preamp (https://tascam.com/us/contents/mic_preamp)
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Tascam products page identifying preamp type by product can be found here: https://tascam.com/us/contents/mic_preamp (https://tascam.com/us/contents/mic_preamp)
So the FR-AV2 has the same ULTRA HDDA preamps as the highly regarded TASCAM HS-P82 field recorder.
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You cannot change the analog input gain on this Tascam! It is fixed!
The ‘record level’ you can set works post ADC. This means it doesn’t really matter what you set it to, you can always normalize it afterwards without any quality impact. (Only if you want to play the recording on the tascam itself, it would be nice to have decent levels which won’t clip during playback.)
Personally I wouldn’t worry at all about the record level and certainly not change it during recording! It won’t bring any benefit recording-quality-wise, and would only complicate things during post processing.
Don't want to make a war out of it, but please read page 10 or 30 in the manual, mind that the Tascam also can record with 24-bit and you need in that case to controle record level BEFORE it hits the ADC. Next, it make no sense in record level behind the ADC, in case of very soft sounds just above the noise floor you never get the bottom out of the first stage bucket when it comes to noise. Due to the nature of maximum levels the preamp is fixed as you suggest.
It’s good to have discussion, and I must admit I assumed recording at 32-bit floating point. However, even when recording at 24 bits it doesn’t necessarily mean there must be analog gain control. The device can operate completely identical regardless recording to 24 bit fixed point or 32 bit floating point storage, apart from that in 24-bit the samples for the output file are converted from the internally used 32bfp to 24bit before storing. And in that case, it would be wise to make sure (by setting the fully digital ‘record level’) that the samples are not too quiet and won’t go over 0dbFS.
It seems Tascam has confirmed there is no analog gain control whatsoever on this device, regardless of the used storage format. See https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100 (https://taperssection.com/index.php?topic=205834.msg2420100#msg2420100).
Of course the dual-ADC setup helps to properly deal with both quiet and loud signals.
A portion from the manual is attached here that kind of confirms too that there is no analog gain control. If there is analog overload, they advice to lower the volume of the source or to move the mics away from the source. They do not tell you to lower the 'record level', as that won't help to relieve the analog part.
Indeed, it was these proposed solutions from this page in the manual that led me to seek clarification from Tascam on whether there was, or was not, gain control in the analog realm. I suppose there is a 3rd option available for those whose mics (like my c480b's) have output pads: enable the pads.
Wouldn't switching to 'line input' practically do the same thing? Of course that would only work for the TRS or XLR inputs, not for the 3.5mm EXT input.
Unless I am not recalling it correctly: unlike the F3, +48v power is not available for 'Line In' mode on the AV2. This difference is where the F3 has an advantage over the Tascam unit, when it comes to using mics that require phantom power.
And correct me if I'm wrong, but the Zoom F3 doesn't even have a 'Line In' mode as such, which would skip the preamp. It's essentially the same analogue path but with a -20 dB pad in front of it. It serves the same purpose when managing levels (i.e. the use case we are discussing here), but the device is not really made for people who for example want to use an external preamp for colouration or less noise.
But then again, I'd think the AV2 would be the same? Line in mode should just be a pad too? Which makes the absence of phantom power a little baffling.
So this is a project or compilation that I'd love to... find, or have someone else do: accurately map or describe the mic in vs line in signal path for a core set of recorders that we use (e.g. SD MixPre# and #ii series, Zoom F-series, FR-AV2, etc...), specifically calling out those that have a separate signal path for line in vs mic in, and those that simply use a pad. I have neither the time nor the technical chops to make meaningful progress on such an initiative.
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I’m fairly sure that the Tascam DR-680 MkI & MkII have separate signal paths for mic & line.
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Recorded another six sets over the last week, SBD / AUD with 2 FR-AV2s. Bluetooth continues to be inexcusably bad (PCM A10 chaining is a must) & I did get another instance of the unclean file splits issue. So it's sporadic. I pressed REC on both devices at the same time, let them hang from there on out, the settings are all the same except for one recording w/ MICs and the other recording LINE IN, and only one of them has the unclean distorted file split so far. About to dig into the other sets soon. Really frustrating, especially given that it isn't consistently reproducible at home. No idea what is possibly causing it, or if it's just total random chance.
(issue being occasionally getting the the tiniest smidgeon of ultra-high-DB-distortion, <.001s, at the end of the 3 hour file, that I am able to manually cut out of my matrixes during the mixdown, but still just extremely annoying)
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I did get another instance of the unclean file splits issue. So it's sporadic.
(issue being occasionally getting the the tiniest smidgeon of ultra-high-DB-distortion, <.001s)
I'm really sorry to hear about your issues. I haven't used my FR-AV2 that much and never experienced the same issues as you. This really sounds like a couple of bytes of non-audio data being interpreted as floating point audio data, resulting in 'random sample values'.
I think I asked before, but I'm not sure about the answer, so I just ask again: What editor (and what version) are you using, and did you already try to open the same original (as in unmodified) file with another editor? Would you be willing and able to share such a faulty file so others can have a look at it? (I'd be happy to give it a try)
It seems that FR-AV2 creates a load of markers at 00:00:00.000 in each file, and maybe some editors won't deal with that properly under certain conditions. (Just speculating... I've seen audio processing software specifically stating they can't handle wav files with more than 'x' markers in it...)
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I did get another instance of the unclean file splits issue. So it's sporadic.
(issue being occasionally getting the the tiniest smidgeon of ultra-high-DB-distortion, <.001s)
I'm really sorry to hear about your issues. I haven't used my FR-AV2 that much and never experienced the same issues as you. This really sounds like a couple of bytes of non-audio data being interpreted as floating point audio data, resulting in 'random sample values'.
I think I asked before, but I'm not sure about the answer, so I just ask again: What editor (and what version) are you using, and did you already try to open the same original (as in unmodified) file with another editor? Would you be willing and able to share such a faulty file so others can have a look at it? (I'd be happy to give it a try)
It seems that FR-AV2 creates a load of markers at 00:00:00.000 in each file, and maybe some editors won't deal with that properly under certain conditions. (Just speculating... I've seen audio processing software specifically stating they can't handle wav files with more than 'x' markers in it...)
It's definitely an issue with the AV2 and I'm pretty (but not 100%) sure there is a tiny bit of actual audio loss along with it. The issue is visible in both Reaper and Audacity, and if it had anything to do with the DAW I'd almost certainly see it across all my split files, not just some of them, as with this run. I'll DM you the raw file.
Since this is a hobby it's like, annoying, but man, if this occurred with a professional shoot, I'd be livid.
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Am I reading that timeline correctly and this is a few 1/1000s of a second?
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Correct
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Interesting. The glitch is exactly 3ms long and positive going. Have you examined any other splits with a glitch, and if so, are they identical to this one? It would be a strong indication of a firmware issue if more users reported this problem. If it’s just your one unit, it could be an individual hardware failure.
This is reminiscent of a problem Sound Devices had with a firmware version for their Mixpre recorders where there was an occasional one sample random (32-bit float) value at the beginning of a file. SD was able to fix the problem with the next update.
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Interesting. The glitch is exactly 3ms long and positive going. Have you examined any other splits with a glitch, and if so, are they identical to this one? It would be a strong indication of a firmware issue if more users reported this problem. If it’s just your one unit, it could be an individual hardware failure.
This is reminiscent of a problem Sound Devices had with a firmware version for their Mixpre recorders where there was an occasional one sample random (32-bit float) value at the beginning of a file. SD was able to fix the problem with the next update.
Good point -- looking back at my original screencap here, it's almost certainly identical, yes. I'll try to be sure to follow through on actually opening a ticket this time.
hm, looks like I'll have another ticket for TASCAM.
What's odd is I didn't notice this in my initial battery tests, I could've sworn I normalized the split tracks during that process but maybe not.
What's attached here is zoomed in waveform at the end of the automatic file split. This is present in both devices I was using. Audible as a tiny "pop" of distortion, normalizes to -INF. When it's simply cut out it is barely noticeable in the recording, but still, should not be occurring.
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I did get another instance of the unclean file splits issue. So it's sporadic.
(issue being occasionally getting the the tiniest smidgeon of ultra-high-DB-distortion, <.001s)
I'm really sorry to hear about your issues. I haven't used my FR-AV2 that much and never experienced the same issues as you. This really sounds like a couple of bytes of non-audio data being interpreted as floating point audio data, resulting in 'random sample values'.
I think I asked before, but I'm not sure about the answer, so I just ask again: What editor (and what version) are you using, and did you already try to open the same original (as in unmodified) file with another editor? Would you be willing and able to share such a faulty file so others can have a look at it? (I'd be happy to give it a try)
It seems that FR-AV2 creates a load of markers at 00:00:00.000 in each file, and maybe some editors won't deal with that properly under certain conditions. (Just speculating... I've seen audio processing software specifically stating they can't handle wav files with more than 'x' markers in it...)
It's definitely an issue with the AV2 and I'm pretty (but not 100%) sure there is a tiny bit of actual audio loss along with it. The issue is visible in both Reaper and Audacity, and if it had anything to do with the DAW I'd almost certainly see it across all my split files, not just some of them, as with this run. I'll DM you the raw file.
Since this is a hobby it's like, annoying, but man, if this occurred with a professional shoot, I'd be livid.
Thanks Kyle for providing the file for analysis.
My initial thoughts were right: This is 'random data' being interpreted as sample data. In fact it's not really random data, but the iXML structure written at the end of the file. This is basically a bunch of ASCII characters containing information about the recording, see attached image '01. Kyle data.png'.
When looking at the structure of your WAV file, I noticed it consists of five parts aka 'chunks' (see attached '02. Kyle chunks.png'):
Chunk 0: The 'bext' chunk, never mind
Chunk 1: The cue points chunk, containing 99 cue points
Chunk 2: The list chunk, containing 99 entries
Chunk 3: The fmt chunk, describing the format (sample rate, #channels etc) of the samples
Chunk 4: The data chunk, containing the actual samples. The iXML data at the end is part of this chunk, so will be interpreted by an editor or player as sample data.
When comparing this against one of my own recordings with the Tascam, I see a sixth chunk at the end, which is the iXML chunk! So the same iXML data is there at the end, but it resides in its own chunk and not in the 'data chunk'. Hence editors will not try to interpret this iXML data as sample data.
So is this a bug? Hell yeah. As it happens sometimes, and not all of the time, I guess this is what they'd call a 'race condition'. Likely two threads of software processing are doing stuff in parallel (e.g. writing the iXML data at the end and then updating the header info about the new sixth chunk in 'the current file' while the other is creating the next file and changing the definition of 'current file' from previous file to new file). Almost always one thread finishes before the other, but occasionally the order of actions reverses, so the software is thinking it is writing to the old file while in fact it is already writing to the new file. Or something like that. I know from experience that multithreading software development can be challenging and it is easy to make errors that just occasionally occur and are difficult to find.
Unfortunately we can't seem to switch off the iXML writing, so that's not an option to prevent this problem from occuring. Just some hints:
- Make sure you run the latest firmware (currently 1.03), as they might already have solved the bug.
- Make sure you format the SD-card in the device, using the devivce formatting routine. Don't format the card from your pc while it is connected to it
Oh, when I just cut off the iXML data using a HEX editor, the file loads just fine without the spikes at the end. I cannot tell/guarantee if there is audio missing or not, as we can't be sure about the effect of the software bug.
Maybe if you'd glue the fixed file and the next file together, you can see if the audio is seamless at the merging point.
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Oh, one addition: As the iXML data in Kyle's file consist of 1168 bytes, and as the sample rate is 48kHz, it represents 3.0234375ms of audio data :)
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Not to try and detract but my buddy's DR-680 mkII occasionally suffers loss on track splits. Seems to be a Tascam issue.
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Not to try and detract but my buddy's DR-680 mkII occasionally suffers loss on track splits. Seems to be a Tascam issue.
I have never had an issue with track splits on the 680 MKII. It might be a particular card or cards to deck issue. Tascam does seem to have issues with cards possibly more than other brands.
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Not to try and detract but my buddy's DR-680 mkII occasionally suffers loss on track splits. Seems to be a Tascam issue.
I have never had an issue with track splits on the 680 MKII. It might be a particular card or cards to deck issue. Tascam does seem to have issues with cards possibly more than other brands.
Card is good and on the list of preferred cards. Tested another card in it, also on that list, and had the same issue. It happens, maybe, once in about 6 or 7 shows especially when running all six channels (including optical out from a V3). But will agree that Tascam decks don't play nicely with a lot of cards.
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Thank you for the phenomenal analysis, Jez. I feel a bit silly for not analyzing the file in a similar manner!
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Thank you for the phenomenal analysis, Jez. I feel a bit silly for not analyzing the file in a similar manner!
No problem whatsoever. I was just fortunate to find a HEX editor that’s able to decode several popular file formats, including WAV files. I thought I would have to decode myself, but this was much easier :) It’s in all our interests to get this issue clear and hopefully solved. I guess it can be helpful to provide the analysis results when reporting to TASCAM…
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I finally received my FR-AV2 yesterday. Very much looking forward to trying it out.
It will be used mostly in a stationary interior installation. Has anyone here powered their unit from a USB wall charger?
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I finally received my FR-AV2 yesterday. Very much looking forward to trying it out.
It will be used mostly in a stationary interior installation. Has anyone here powered their unit from a USB wall charger?
Yes. It worked fine using a typical 5V USBA to USBC cable and a little AC adapter.
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I finally received my FR-AV2 yesterday. Very much looking forward to trying it out.
It will be used mostly in a stationary interior installation. Has anyone here powered their unit from a USB wall charger?
Yes. It worked fine using a typical 5V USBA to USBC cable and a little AC adapter.
Great! Thanks!
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Just taken delivery of my Tascam FR-AV2 and been playing with it. Some early observations.
It came with the original firmware which I have to say really was not fit for purpose. Having to input the settings every time the unit was switched on is a PITA. Sorted after the firmware upgrade, but it makes me realise what some have already said about manufacturers releasing products and letting the customers test them.
Each time I change input settings it allows me to save as one of three Presets. I have no idea how to choose one of those Presets. :banging head:
When using bluetooth I am getting some interference on some microphones making them unusable. Maybe it’s the microphones, but they work okay with my Mixpre 3 when using bluetooth.
Other than that I am pretty happy with the unit so far.
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Just to add my experience with battery life. Two Sennheiser MKH 8040 microphones, 48v power, three freshly charged Ikea batteries, no power save options so screen on continuously … 7 hours 23 minutes.
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I have figured how to save input settings as presets, and where to load them, but cannot get this to work.
I want to save the settings for XLR mics with 48v power as Preset 1, and plug in mics with 5v PIP as Preset 2. But every time I load Preset 2 I just get the same settings as Preset 1. Anyone else getting success with this? I have updated to the latest firmware.
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I have figured how to save input settings as presets, and where to load them, but cannot get this to work.
I want to save the settings for XLR mics with 48v power as Preset 1, and plug in mics with 5v PIP as Preset 2. But every time I load Preset 2 I just get the same settings as Preset 1. Anyone else getting success with this? I have updated to the latest firmware.
Having actually read the manual now I know why I can’t get the presets to save input type and power, it’s because it doesn’t actually claim to. It can save all settings but these. Unfortunately these are the main settings I want saving so it seems I will just have to change each time I switch between 48v and 5v mics. Not a deal breaker but slightly annoying.
What’s more annoying is that I simply cannot get any bluetooth headphone to wirelessly monitor. As others have mentioned the Bluetooth seems to be very unstable, and I’m wishing I didn’t buy the AK-BT2 adapter.
Other than that I am liking the unit, the size the functionality and above all the sound quality of the recordings.
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Just finding this thread. I picked up an FR-AV2 with the intention of using it for stealthing with a pair of MK4>CMC1L. I recorded my first show with it and the audio sounds great; on par with my MixPre6 II. I didn't get the bluetooth dongle in time so was only able to interact on the device itself, which was a pain since feeling around for buttons in a pocket isn't useful when all the buttons are the same shape/size.
Overall, the unit is relatively large. Not exactly the most discrete device.
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Just finding this thread. I picked up an FR-AV2 with the intention of using it for stealthing with a pair of MK4>CMC1L. I recorded my first show with it and the audio sounds great; on par with my MixPre6 II. I didn't get the bluetooth dongle in time so was only able to interact on the device itself, which was a pain since feeling around for buttons in a pocket isn't useful when all the buttons are the same shape/size.
Overall, the unit is relatively large. Not exactly the most discrete device.
How did you set up the mics? Thank you.
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I just recorded a classical guitarist with this recorder and an m/s pair of Sennheiser mkh 8040/8030 mics.
The recording sounds stupendous. The recorder allows the option to monitor in m/s, a big advantage over
the Zoom F3 for m/s recordists.
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I just recorded a classical guitarist with this recorder and an m/s pair of Sennheiser mkh 8040/8030 mics.
The recording sounds stupendous. The recorder allows the option to monitor in m/s, a big advantage over
the Zoom F3 for m/s recordists.
Nice!
Hopefully Zoom takes notice and modifies M/S monitoring and playback options on their F-series recorders with firmware updates to provide adjustability of Mid/Side ratio and the level of the M/S pair in the mix.
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I just recorded a classical guitarist with this recorder and an m/s pair of Sennheiser mkh 8040/8030 mics.
The recording sounds stupendous. The recorder allows the option to monitor in m/s, a big advantage over
the Zoom F3 for m/s recordists.
Nice!
Hopefully Zoom takes notice and modifies M/S monitoring and playback options on their F-series recorders with firmware updates to provide adjustability of Mid/Side ratio and the level of the M/S pair in the mix.
Agreed!
Admittedly the v2.0 firmware for the F3 is mature and stable, but I think Zoom is missing out on sales now due to the lack of Mid-Side support. Even just monitoring would be nice. I still use my Mid-Side mic rig with my F3, though, despite the inconvenience. A firmware feature update is long overdue.
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The F6 and the F8 variants do have good m/s capabilities and routing features. They should put it in the F3
since a lot of video and films sound people could use that feature.
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And in the H1 XLR! Its menu system is so similar to the H4e but it strangely lacks the MS bit totally. I have a Sennheiser MS mic preamp which I bought in the mid eighties - kinda rare beast now - can run on batteries but it's bigger than the H1 XLR!
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The F6 and the F8 variants do have good m/s capabilities and routing features. They should put it in the F3
since a lot of video and films sound people could use that feature.
Don't want to go too OT in this FR-AV2 thread and not sure about F6, but although F8 is able to monitor M/S, its implementation is weak, with no adjust Mid/Side ratio nor the level of the M/S pair in the mix
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I have a FR-AV2 for sale in the Yard if anyone is looking.
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Anyone using this with a Babynbox? I'm not sure which input source to use: EXT or Line. Don't seem to be getting a signal on Line.
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Anyone using this with a Babynbox? I'm not sure which input source to use: EXT or Line. Don't seem to be getting a signal on Line.
If you connect your source to the XLR/TRS inputs, you must choose MIC or LINE, depending on the input level. If you connect your source to the EXT mini-jack input, you must select EXT.
In all my ignorance, I'm not sure why you'd need the Babynbox. The Tascam has an excellent pre-amp and can deliver 5V pip on the EXT input. If that is sufficient for your mics then you might be able to drop the NBox...
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^ Nbox/babyNbox provides 1) high polarization voltage for capsules and 2) unbalanced output, allowing use into small recorders that don't have phantom power and XLR input. It replaces the microphone amplifier bodies which require phantom power and produce a balanced output. Could alternately use standard mic-bodies into the XLR inputs of FR-AV2. But this solution makes for a very compact stereo mic arrangement by eliminating the mic bodies.
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I hear ya'll on not needing the Babynbox but I'm coming from a stealth setup of MK4/MK41 > Babynbox > PCM-A10 so swapping out the A10 with the FR-AV2 is a good first step. I'm intrigued by the idea of swapping out the Babynbox with a set of CMCs but I'm looking at BH&H (always dangerous) and it looks like a compact set of CMCs start around 1K each, specifically the CMC1 series like the CMC1L. Depending on that configuration you'd need to either go MK4 > CMC -> Cables -> FR-AV2 which might get clunky in a hat or MK4 > Cables -> CMC -> FR-AV2 which would require the KCs and those are like another $400 each. Those cables must be made from the blood of christ or something because that seems insanely expensive. But minimally, unless something popped up in the yard sale, that's another 2K just to replace the Babynbox+cables and I don't want to drop that kind of cash just yet.
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swapping to kc cables and cmc6 bodies would also make the rig bigger and more challenging to get into a venue
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I hear ya'll on not needing the Babynbox but I'm coming from a stealth setup of MK4/MK41 > Babynbox > PCM-A10 so swapping out the A10 with the FR-AV2 is a good first step. I'm intrigued by the idea of swapping out the Babynbox with a set of CMCs but I'm looking at BH&H (always dangerous) and it looks like a compact set of CMCs start around 1K each, specifically the CMC1 series like the CMC1L. Depending on that configuration you'd need to either go MK4 > CMC -> Cables -> FR-AV2 which might get clunky in a hat or MK4 > Cables -> CMC -> FR-AV2 which would require the KCs and those are like another $400 each. Those cables must be made from the blood of christ or something because that seems insanely expensive. But minimally, unless something popped up in the yard sale, that's another 2K just to replace the Babynbox+cables and I don't want to drop that kind of cash just yet.
You could see about getting your nbob cables terminated with Naiant PFAs
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What cards are everyone running? From the approved list?
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fwiw, i was finally able to find the line of Kioxia approved cards on amazon japan with free international shipping right now
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What cards are everyone running? From the approved list?
I'm using
https://www.amazon.com/dp/B09X7CRKRZ?ref_=ppx_hzsearch_conn_dt_b_fed_asin_title_1&th=1
SanDisk 256GB Extreme microSDXC UHS-I Memory Card with Adapter - Up to 190MB/s, C10, U3, V30, 4K, 5K, A2, Micro SD Card - SDSQXAV-256G-GN6MA
No issues thusfar
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I’ve been using my Tascam FR-AV2 as a portable recorder with pluggy mics directly into the XLR inputs.
Unfortunately Tascam has Inputs 1 (Left) and 2 (Right) positioned differently to Zoom, and screen up the mic pointing right is the left channel, and the mic pointing left is the right channel. I don’t really want to be fixing later so have been placing the recorder face down, which is not ideal. I’ve read the manual to determine if there is a way to switch channels but can’t see anything. I would be grateful if someone can tell me if I’m missing something obvious.
I notice that Immersive Soundscapes have a new product that is very interesting. I have some MKH8040 mics and this could be the ultimate portable recorder. It’s just the channels the wrong way round that’s a bit annoying.
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Can't you just rotate it 180º?
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Can't you just rotate it 180º?
Yes, that’s putting the recorder face down on the surface.
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Can't you just rotate it 180º?
Yes, that’s putting the recorder face down on the surface.
Well, then flip it 180º the other way.. :bigsmile:
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Gotcha. I kind of misinterpreted your previous post, as I was looking at the orientation in the photo. Obviously, though, you would be using it horizontally and not vertically...
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Gotcha. I kind of misinterpreted your previous post, as I was looking at the orientation in the photo. Obviously, though, you would be using it horizontally and not vertically...
Gotcha too. I didn’t know if I was being dumb.
Actually vertical orientation does make sense as upwards facing inputs gives a lot more options with the adapters available. I’ve considered it, with some sort of stand attached, but it would be a lot less stealth like and I like to make recordings unnoticed.
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I've been testing the FR-AV2 with a few different mics having sensitivity ranges from 8 mv/pa to 1.2 mv/pa to determine level setting characteristics and the +4 db max input value stated by Tascam.
I tested in an outdoor concert environment and the source was a Martin Audio Wavefront line array pushing sound pressures at the mic position averaging 80db with peaks in the low 90db range.
Here are my findings:
With the most sensitive condenser mics I own (8 mv/pa) the max input gain setting to get good levels without exceeding 24bit full scale was +8 to +10 db.
With the least sensitve mic I own (dynamic) the gain setting on the Tascam averaged 28-30db.
In summary, mics with a sensitivity spec at 10 mv/pa will only allow 5-6 db gain setting on the recorder. Very sensitive mics at 15 mv/pa or greater will require the gain on the Tascam to be set to minimum.
The sweet spot mic sensitivity for concert taping with the FR-AV2 seems to be in the 2.5 to 3.5 mv/pa range. You can use even less sensitive mics (1-2 mv/pa) but I would stay away from mics with higher sensitivity specs above 10-15 mv/pa.
Hope this info is helpful in selecting the right microphones to use with the FR-AV2 for concert taping.
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I've been testing the FR-AV2 with a few different mics having sensitivity ranges from 8 mv/pa to 1.2 mv/pa to determine level setting characteristics and the +4 db max input value stated by Tascam.
<snip>
Hope this info is helpful in selecting the right microphones to use with the FR-AV2 for concert taping.
Forgive me if I'm wrong, but... I think it was determined before that the FR-AV2 actually has a fixed analog gain (regardless of the 24/32bfp recording format), and the gain setting done by the user is applied in the digital domain (after the ADC).
If you definitely want to record straight to 24-bit for whatever reason, then for sure you want to make sure you won't set this digital gain so high that the output won't fit into the 24-bit dynamic range. In that situation your test is relevant.
However, running the risk of going beyond the 24-bit dynamic range can be mitigated completely by simply recording to 32-bit float. It will not improve or reduce the sound quality (*1), it will simply prevent overloading your output format regardless of your (digital) gain setting. During post-processing you can normalize and export to 24-bit. So why run the risk of format-overloading in the first place?
I guess it would be more interesting to find out at what sound pressure the analog stage overloads for different sensitivity mics... (User manual: "If an overload occurs with an analog circuit, the entire level meter will become red.") I'd like to believe that setting the digital gain to 0dB would give a nice impression of that (meaning 0dB on the VU level meters is on the edge of overloading the analog stage), but I haven't been able yet to test if this is actually true or not due to lack of loud enough sounds...
1) This is a bit debatable, but that's a different discussion. I think we can say though that storing in 32bfp could only make things better (especially for quiet parts in your recordings) and certainly not worse.
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I've been testing the FR-AV2 with a few different mics having sensitivity ranges from 8 mv/pa to 1.2 mv/pa to determine level setting characteristics and the +4 db max input value stated by Tascam.
<snip>
Hope this info is helpful in selecting the right microphones to use with the FR-AV2 for concert taping.
Forgive me if I'm wrong, but... I think it was determined before that the FR-AV2 actually has a fixed analog gain (regardless of the 24/32bfp recording format), and the gain setting done by the user is applied in the digital domain (after the ADC).
If you definitely want to record straight to 24-bit for whatever reason, then for sure you want to make sure you won't set this digital gain so high that the output won't fit into the 24-bit dynamic range. In that situation your test is relevant.
However, running the risk of going beyond the 24-bit dynamic range can be mitigated completely by simply recording to 32-bit float. It will not improve or reduce the sound quality (*1), it will simply prevent overloading your output format regardless of your (digital) gain setting. During post-processing you can normalize and export to 24-bit. So why run the risk of format-overloading in the first place?
I guess it would be more interesting to find out at what sound pressure the analog stage overloads for different sensitivity mics... (User manual: "If an overload occurs with an analog circuit, the entire level meter will become red.") I'd like to believe that setting the digital gain to 0dB would give a nice impression of that (meaning 0dB on the VU level meters is on the edge of overloading the analog stage), but I haven't been able yet to test if this is actually true or not due to lack of loud enough sounds...
1) This is a bit debatable, but that's a different discussion. I think we can say though that storing in 32bfp could only make things better (especially for quiet parts in your recordings) and certainly not worse.
I'm not sure where I claimed the preamp gain control was analog, because it's not, and it's fixed. The input level max of +4 would be difficult to reach, even for sensitive microphones. A Schoeps CMC6/MK4 combo would need 130db of sound pressure to generate enough voltage to hit that max, but that's not the point. Because that mic's sensitivity is 15 mv/pa, in a concert setting you would have to reduce gain to minimum and you are correct that 32bit float will handle signals that would otherwise clip in 24bit, but I didn't want a brick that I can't control levels, specificallly because I can't stand uneven channels between left and right.
I should have started out by explaining that I don't give a rip about 32bit float for recording a concert PA where the dynamic range of the music is no more than 60db. I ran 32bit float once and I hated the workflow so much that I doubt I'll ever use it.
I've spent the last 32 years managing digital recorders at shows where the top of the scale for levels is zero from the DAT days until now. I'm sure young tapers who haven't had such a long period of muscle memory development are baffled by the position that going over zero is never ok.
I am hard wired that the measure of sucess recording shows is how little post work is required, which I know is very old thinking but I won't be changing that now. I am too old.
My tests were to find the sensitivity value that will allow me to use and adjust gain to avoid post work. Thanks
(Edit to add "preamp" to sentence about analog gain)
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This looks interesting!
https://reverb.com/item/85619654-earsight-the-one-for-tascam-fr-av2-stereo-field-recording-microphone
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I've been testing the FR-AV2 with a few different mics having sensitivity ranges from 8 mv/pa to 1.2 mv/pa to determine level setting characteristics and the +4 db max input value stated by Tascam.
<snip>
Hope this info is helpful in selecting the right microphones to use with the FR-AV2 for concert taping.
Forgive me if I'm wrong, but... I think it was determined before that the FR-AV2 actually has a fixed analog gain (regardless of the 24/32bfp recording format), and the gain setting done by the user is applied in the digital domain (after the ADC).
If you definitely want to record straight to 24-bit for whatever reason, then for sure you want to make sure you won't set this digital gain so high that the output won't fit into the 24-bit dynamic range. In that situation your test is relevant.
However, running the risk of going beyond the 24-bit dynamic range can be mitigated completely by simply recording to 32-bit float. It will not improve or reduce the sound quality (*1), it will simply prevent overloading your output format regardless of your (digital) gain setting. During post-processing you can normalize and export to 24-bit. So why run the risk of format-overloading in the first place?
I guess it would be more interesting to find out at what sound pressure the analog stage overloads for different sensitivity mics... (User manual: "If an overload occurs with an analog circuit, the entire level meter will become red.") I'd like to believe that setting the digital gain to 0dB would give a nice impression of that (meaning 0dB on the VU level meters is on the edge of overloading the analog stage), but I haven't been able yet to test if this is actually true or not due to lack of loud enough sounds...
1) This is a bit debatable, but that's a different discussion. I think we can say though that storing in 32bfp could only make things better (especially for quiet parts in your recordings) and certainly not worse.
I'm not sure where I claimed the gain was analog, because it's not. The input level max of +4 would be difficult to reach, even for sensitive microphones. A Schoeps CMC6/MK4 combo would need 130db of sound pressure to generate enough voltage to hit that max, but that's not the point. Because that mic's sensitivity is 15 mv/pa, in a concert setting you would have to reduce gain to minimum and you are correct that 32bit float will handle signals that would otherwise clip in 24bit, but I didn't want a brick that I can't control levels, specificallly because I can't stand uneven channels between left and right.
I should have started out by explaining that I don't give a rip about 32bit float for recording a concert PA where the dynamic range of the music is no more than 60db. I ran 32bit float once and I hated the workflow so much that I doubt I'll ever use it.
I've spent the last 32 years managing digital recorders at shows where the top of the scale for levels is zero from the DAT days until now. I'm sure young tapers who haven't had such a long period of muscle memory development are baffled by the position that going over zero is never ok.
I am hard wired that the measure of sucess recording shows is how little post work is required, which I know is very old thinking but I won't be changing that now. I am too old.
My tests were to find the sensitivity value that will allow me to use and adjust gain to avoid post work. Thanks
I didn't mean to step on any toes, sorry if I did. All respect for the tests you did and sharing the results.
IMHO there are few reasons why one would opt for 24bit storage if you'd have a (proper multi-ADC) 32bfp recorder in your hand, and 'hating the 32bit float workflow' sure sounds like a valid one :). It does make me curious what it is that you hate so much about the 32bit workflow. Personally I don't experience much difference from the 24bit workflow at all as I've almost always been normalizing anyway since I replaced my WMD-6C with a portable DAT 32 years ago. I just love the thought that I don't have to worry at all about clipping or underwhelming during recording and editing. I don't mind the 33% extra storage space and transfer times for this benefit. To me, recording in 24bit on a 32bfp recorder feels a bit like switching off the airbags in the car because one's trying to avoid collisions anyway. To each their own, obviously!
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This looks interesting!
https://reverb.com/item/85619654-earsight-the-one-for-tascam-fr-av2-stereo-field-recording-microphone
That's neat, but the cost is a bit scary - current price of the Zoom M4 is about the same, recorder included. And in the context of Taperssection, the M4 handles 135db SPL, 15dB more than this mic. But I guess a side by side comparison would be required to really tell the difference and - that ain't going to happen.
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Digital clipping has never been a pain point for me. I guess setting levels is something I've gotten down fairly well after some unknown number in the hundreds of shows I have taped. I should be a huge 32bit float guy because I like to run hot which I guess is a leftover habit from the analog cassette days when saturation was the key to minimizing tape noise. If a recording has a dozen clipped sectors, the only way I can tell is when I see it on a DAW. I sure can't hear it, and I don't know anyone else who has either.
About the 32 bit workflow. I should have said I don't like the required workflow of that format. Right in the venue parking lot I can take a 24/48 raw master file, copy it over to a usb c thumbdrive, and plug that into the usb slot on the head unit of my car stereo and have it playing in 5 minutes.
Sorry, but I think the airbag analogy is ridiculous. A better analogy is the Canon A-1 SLR camera that was introduced in 1978. That was the first professional 35mm camera to feature a fully automatic exposure system. There were already cameras with either aperture priority or shutter priority, but the A-1 was the milestone that evolved to the "anybody can take a good photo" goal of all photography equipment after that. Of course, composition and balance and artistic qualities still had to come from the skill of the photographer, so when the masses could run out and get a fully auto slr by the early 1980s, suddenly there were a million amateur photographers still taking shitty photos, but now they were not over or under exposed. Even after Nikon, Minolta, and Pentax jumped on the automatic camera bandwagon, their top line professional cameras could still be switched to full manual with a good old center weighted match needle light meter.
Why would they make cameras with fully manual controls if automatic cameras could that work better. Well, it's that the "better" part only applied to amature weekend snapshot types, pros knew how to select shutter speeds and apertures using a hand held light meter and their industry was not screaming "help we need an automatic camera to make all the settings for us". The only reason automatic exposure systems were found in the top end cameras afterwards was purely for the purpose of selling them to pro wannabes who had no clue how that stuff worked. Same thing happened a decade later with auto focus. No pro ever said "I wish I didn't have to focus manually". 32bit float is a cool feature, but it's not to someone who rarely if ever needed to have, IMO of course.
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With photography, it's artistic choices that are enabled by using manual mode. You can do something different than the system is going to do, so your pictures are demonstrably different, and that can't be fixed in post.
With 32 bit audio vs 24 bit, using just digital gain, once normalized both files are identical, unless you got clipping in 24 bit. So you lose by using 24 bit, rather than gain. But that said, I usually record in 24 bit because that's what the gear I use in the field does, unless I'm bringing the big recorder.
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This looks interesting!
https://reverb.com/item/85619654-earsight-the-one-for-tascam-fr-av2-stereo-field-recording-microphone
Dig that all-in-one with a removable baffle idea. 3D printing nicely applied, as seems to be the case in a number of the Immersive Soundscape products. Should be relatively easy for DIY'ers with 3D printing skills to emulate.
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Or maybe support the guy who developed and implemented the idea...
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Sure. Implemented at least. Old idea we were DIY'ing around here at TS 20 years ago.
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... I usually record in 24 bit because that's what the gear I use in the field does, unless I'm bringing the big recorder.
I usually record in 32 bit because that's what the gear I use in the field does, unless I'm bringing the big recorder. :) In my case the choice would probably lie between the Zoom H1 XLR and the dear old Tascam 60D - the former of course being radically smaller than the latter.
Actually, these days if I had to make a paid-for classical music recording like in the good old days, when it often just needed a Sennheiser MS pair in the right spot, I would connect the mics to the the Zoom, then connect the output thereof to the Tascam (because standard practice in the old days was to use two recorders for overall backup). The Zoom annoyingly has no MS settings, but the Tascam does, so for replay to artists I could use the Zoom but listen to it through the Tascam, with its MS monitoring mode. You can even screw the Zoom on top of the Tascam to make a kind of one-piece device. But I digress - again...
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I’d use the Tascam as the preamp and the zoom for the recording in that scenario. For me the lower noise floor matters most.
I use either the dpa dvice into my phone or the deity pr2 for most situations and the Sonosax r4+ when I can bring the big rig.
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I’d use the Tascam as the preamp and the zoom for the recording in that scenario. For me the lower noise floor matters most.
The Tascam DR-60D (Mk I which I have, or the later Mk II) noise spec is -120dBu or less, while the Zoom H1 XLR is -122, so the Zoom has a slight advantage (partly I guess because it's about ten years younger!). I do think that Zoom is a bit of an under-rated product, and maybe using the H1 designation was a mistake, as it makes people expect it to be noisy, and it isn't. Of course the FR-AV2 noise level is -127dB, for which you pay!
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Sorry, but I think the airbag analogy is ridiculous. A better analogy is the Canon A-1 SLR camera that was introduced in 1978. That was the first professional 35mm camera to feature a fully automatic exposure system. There were already cameras with either aperture priority or shutter priority, but the A-1 was the milestone that evolved to the "anybody can take a good photo" goal of all photography equipment after that. Of course, composition and balance and artistic qualities still had to come from the skill of the photographer, so when the masses could run out and get a fully auto slr by the early 1980s, suddenly there were a million amateur photographers still taking shitty photos, but now they were not over or under exposed. Even after Nikon, Minolta, and Pentax jumped on the automatic camera bandwagon, their top line professional cameras could still be switched to full manual with a good old center weighted match needle light meter.
Why would they make cameras with fully manual controls if automatic cameras could that work better. Well, it's that the "better" part only applied to amature weekend snapshot types, pros knew how to select shutter speeds and apertures using a hand held light meter and their industry was not screaming "help we need an automatic camera to make all the settings for us". The only reason automatic exposure systems were found in the top end cameras afterwards was purely for the purpose of selling them to pro wannabes who had no clue how that stuff worked. Same thing happened a decade later with auto focus. No pro ever said "I wish I didn't have to focus manually". 32bit float is a cool feature, but it's not to someone who rarely if ever needed to have, IMO of course.
Hmm, well, I guess when using the camera analogy, then automatic exposure seems to me more like the auto gain control or limiter. Then everybody can make recordings without clipping. Going to 32bfp seems like more like they finally invented a film or camera sensor with such a high dynamic range that you can take photos where shutter time and amount of light don't matter anymore. Even the brightest lights with extremely long exposure times won't over-expose, and the darkest scenes can be shot with very short exposure times and still capture all the details in the dark. Only when you decide to print it on paper with its limited dynamic range (i.e. convert your audio to 24 or 16 bits), you have to determine what to do with the shot (i.e. compress, normalize, whatever) to fit it in the limited dynamic range of the target. There are no real artistic considerations regarding setting the record level, its just something you have to do 'good enough for the circumstances', which can be skipped completely when recording to 32bfp. Other stuff (getting your stealth equipment in, positioning, mic orientation etc) I do like to consider a work of art :bigsmile:
Don't get me wrong: I would have loved to have a recorder that can record in 32bfp yet still allows to set the analog gain! It would re-introduce the risk of clipping in the analog domain (and therefor re-introduce the 'art of setting the record level right'), but also allows to optimize the gain stage for lower noise. :cheers:
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My short way of arguing the merits of 32 bit float (which I'm sure we have discussed to death in these forums over the years anyway) is this - if we'd had 32 bit float from the outset, and then one of the manufacturers suddenly brought out the first ever 24 bit audio recorder together with new fangled gain controls to make sure we didn't clip stuff etc, would we all have rushed to buy it, on the grounds that it would be a better way of doing things? Personally I don't think so, but of course some might disagree!
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The value of these recorders is not in the 32-bit floating point aspect, which is just a convenient storage format, but the use of multiple ADCs. Multi-ADC designs in 24-bit, such as the original MixPres, offer pretty much the same advantages (wide dynamic range).
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Sure. Implemented at least. Old idea we were DIY'ing around here at TS 20 years ago.
Exactly....just like the birth of your oddball microphone configuration DIY project.
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My short way of arguing the merits of 32 bit float (which I'm sure we have discussed to death in these forums over the years anyway) is this - if we'd had 32 bit float from the outset, and then one of the manufacturers suddenly brought out the first ever 24 bit audio recorder together with new fangled gain controls to make sure we didn't clip stuff etc, would we all have rushed to buy it, on the grounds that it would be a better way of doing things? Personally I don't think so, but of course some might disagree!
Totally agree
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Exactly....just like the birth of your oddball microphone configuration DIY project.
Well, to be fair, the Immersive Soundscapes product, containing, as it does, mics, a 3-D printed body, and Neutrik connectors, is a pretty far cry from taping a piece of cardboard in between your handheld's on-board mics. And baffling microphones has been a thing for, what, 100 years? That didn't arise from taperssection.com...
Immersive also has an ORTF version with cardioids. And they are available for the MixPre, PMD661 (MKII), and F3. They both spec a max dBSPL of 110 dB, though.
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Exactly. Not a new idea - neither here or there.
Long live DIY.. AND long live the folks who provide gear for others who don't have the desire or skill to do it themselves. Fortunately there's room in the world for both. As well as a long history of support for both in the history of amateur concert taping.
Edit- ..and those two worlds converge in interesting ways. A key demographic here at TS has long been members who build things for other tapers. Huge thanks to them. Immersive Soundscapes extends that in an interesting way by making available products well within the reach of some DIYers thanks to the development of 3d printing which supports both DIY and low-quantity commercial production efforts. Win win, and a cool convergence. I'm all in for the consolidation of good ideas and the implementation of them by both means when the result is better recordings.
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I see how my photography analogy fell apart due to the complexities that differ between the two technologies. The intent of my first post was merely to share my experiences with different sensitivity microphones through the FR-AV2. After reading through this thread again, there is definitely a "set and forget" camp standing strongly behind 32bit float as the answer to that need.
Here is what I know. Max input specs on recorders are based on gain set to minimum. If you use a hot mic, 10mv/pa to 15mv/pa, you will have to turn the gain down all the way at a rock show. No control of balance, and no real sense of headroom. However, if you run a low sensitivity condenser or a dynamic microphone and are able to apply -any- gain, you know that your input levels are unlikely to push past the +4 threshold. So it doesn't matter if the gain is analog, digital, or a combination, being able to add 15-20db of gain tells you the incoming signal from the mic has near zero chance of overloading the recorder mic inputs. It also doesn't matter if you are running 24bit or 32bit, the benefits of having more control are the same. One thing I have learned over the years with recorders is they all have an optimal gain level. I think the optimal level for this recorder is revealed by the +18db factory default gain setting. The thing that's great about this recorder is it's broad feature set and applicability across wide array of uses and users.
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Don’t know if this has been mentioned previously but just been out to do some comparison field recordings with my Sony PCM-D100 and Tascam FR-AV2, it was a sunny day and the screen on the AV2 was absolutely impossible to read making it unusable. The simple LCD screen on the D100 was perfectly clear.
It doesn’t seem to take much ambient brightness to really hinder this screen.
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Don’t know if this has been mentioned previously but just been out to do some comparison field recordings with my Sony PCM-D100 and Tascam FR-AV2, it was a sunny day and the screen on the AV2 was absolutely impossible to read making it unusable. The simple LCD screen on the D100 was perfectly clear.
It doesn’t seem to take much ambient brightness to really hinder this screen.
You can adjust the display contrast in the menu but not the brightness IIRC. Did you try that to see if it was easier to read?
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You can adjust the display contrast in the menu but not the brightness IIRC. Did you try that to see if it was easier to read?
Good point. I’ve just checked and I had it set to ‘mid’. I’ll check the next sunny day (in the UK so might be a while) and set to ‘high’.
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Don’t know if this has been mentioned previously but just been out to do some comparison field recordings with my Sony PCM-D100 and Tascam FR-AV2, it was a sunny day and the screen on the AV2 was absolutely impossible to read making it unusable. The simple LCD screen on the D100 was perfectly clear.
It doesn’t seem to take much ambient brightness to really hinder this screen.
I had major issues with the FR-AV2 in the bright sun at Rooster Walk. The screen was almost impossible to read. I got lucky with the recording settings and turning phantom power on/off. Yesterday, I tested every possible option of Brightness and Contrast in the bright sun, to no avail. I have zero issues with the sound devices units or the Zoom F6 in the bright sun. I wonder if a software update in the future could solve this one?
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I wonder if a software update in the future could solve this one?
It sounds like a physical characteristic of the used display and/or coating/protection on top of it, so I really doubt any software change could substantially improve it.
A possible work-around could be to use the Bluetooth dongle and then control the device from your phone. That might have better readability in bright sunlight.
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It was sunny again today so checked again with the brightness set to ‘high’. It was marginally easier to read the screen, but still very difficult.
I’m aware that I can use the bluetooth function, but would prefer not to. I prefer the simplicity of using the controls on the unit itself, and I have a set of mics that pickup interference when using bluetooth.
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Maybe try one of the anti-glare screen protectors? I have never tried one, but they are advertised specifically for use in sunny conditions.
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I was lurking on Amazon today and ran across this:
https://www.amazon.com/OPENMOON-Inch-Monitor-Hood-Sunshade/dp/B07BPWVBQS
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Finally managed to acquire some IK Multimedia iRig XY Microphones. This might just be my perfect portable field recorder. They just fit into the Tascam FR-AV2 in an XY position. This is useful in the AV2 because the XLR input on the right is for the left channel.
I look forward to doing some comparisons with my Sony D100 and other recorders and microphones.
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Finally managed to acquire some IK Multimedia iRig XY Microphones. This might just be my perfect portable field recorder. They just fit into the Tascam FR-AV2 in an XY position. This is useful in the AV2 because the XLR input on the right is for the left channel.
I look forward to doing some comparisons with my Sony D100 and other recorders and microphones.
Do you have any idea what mic capsules are in the iRig Mic XY? 22db self noise Cardioids seems a bit high. Thanks.
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Do you have any idea what mic capsules are in the iRig Mic XY? 22db self noise Cardioids seems a bit high. Thanks.
The capsules are visible since the tops are removable, but no idea what they are. They don’t seem particularly noisy on initial tests, but will do some comparisons over the weekend.
I think it would be easy to switch for some superior capsules if I feel the need to. Although the current capsules are 13.8mm and the internal diameter of the housing is 14.2mm so the Primo EM200 (14.6mm) are probably not an option.
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Do you have any idea what mic capsules are in the iRig Mic XY? 22db self noise Cardioids seems a bit high. Thanks.
The capsules are visible since the tops are removable, but no idea what they are. They don’t seem particularly noisy on initial tests, but will do some comparisons over the weekend.
I think it would be easy to switch for some superior capsules if I feel the need to. Although the current capsules are 13.8mm and the internal diameter of the housing is 14.2mm so the Primo EM200 (14.6mm) are probably not an option.
Thanks for the look inside. I had never heard of this company until you posted this.
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Thanks for the look inside. I had never heard of this company until you posted this.
I’ve had great benefits from their iRig Pre Hd which allowed me to attach a xlr P48 mic to my iPhone with very little noise (mono).
An alternative to the IK Multimedia mics could be those offered by Centrance PivoMic PM1. They are referring to a Japanese supplier of the capsules so I am guessing PRIMO. Kind of expensive tough.
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The problem with the Centrance PivotMic is that the XLR connectors are oriented weirdly, so not really suitable for the Tascam AV2 or the Zoom F3. Also expensive and not available.
I really don’t understand why someone hasn’t come up with a quality product like this, there seems to be so much pent up demand online.
My hobby is stealth recording ambiences, usually spontaneously and frequently with people around. I usually use the Sony D100 which I can set down unnoticed. But it is really limited in other ways, and very susceptible to wind noise.
I really think the Tascam AV2 has potential to be the perfect portable field recorder. Set up as a portable for immediate use like the D100 but flexible enough to do so much more with it when the situation allows.
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The problem with the Centrance PivotMic is that the XLR connectors are oriented weirdly, so not really suitable for the Tascam AV2 or the Zoom F3. Also expensive and not available.
I really don’t understand why someone hasn’t come up with a quality product like this, there seems to be so much pent up demand online.
My hobby is stealth recording ambiences, usually spontaneously and frequently with people around. I usually use the Sony D100 which I can set down unnoticed. But it is really limited in other ways, and very susceptible to wind noise.
I really think the Tascam AV2 has potential to be the perfect portable field recorder. Set up as a portable for immediate use like the D100 but flexible enough to do so much more with it when the situation allows.
I use the Centrance Pivot Mics with the Zoom F3 and Tascam AV2 often. I just put either deck on it's side. There also is a product (I forget the name or manufacturer) that I have the allows me to extend and orient the mics any way I want, but it makes the overall unit larger much like the Immersive sound earsight mics, so I use it only sometimes. Turning it on it's side works great and there is no phase issues as one mic is higher than the other.
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Any recent experiences powering DPA 4060 with this unit? Will the onboard power do the job?
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The PIP is 5V so it should work just dandy. That said, I've only verified function with my 4015gs, and haven't used it in the field yet.
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Any recent experiences powering DPA 4060 with this unit? Will the onboard power do the job?
I’ve powered my DPA 4060 and 4061 mics just fine with this recorder.
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Thanks for the responses.
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I’ve powered my DPA 4060 and 4061 mics just fine with this recorder.
Are you happier with the quality of your recordings on the FR-AV2 than on your MixPre 3 ii ?
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Are you happier with the quality of your recordings on the FR-AV2 than on your MixPre 3 ii ?
I can’t say I’ve noticed much difference in sound quality, but haven’t done a direct comparison. Maybe the MixPre 3 has quieter amps.
I am considering selling the MixPre 3 ii though for the following reasons:
1. Size and convenience. I can go into the field with mics plugged in directly into the AV2 and use as a ‘portable all in one’ recorder, and add cables and better mics if I need.
2. Battery life on the AV2 is much better.
3. The AV2 is much simpler to use. The learning curve of the MixPre 3 is steep and I find myself having to revisit the instructions each time I use it.
4. I prefer the files directly off the AV2.
5. The 2.5v PIP on the MixPre 3 has always irritated me.
These reasons could apply to the Zoom F3 which I have never used.
The only thing stopping me from selling the MixPre is that the screen is easier to use, the mounting points top and bottom is useful, and the fact that I paid to have the additional two channels and have not used them yet. The main reason though is that I think it’s a beautiful bit of kit and I don’t need the money!!
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Are you happier with the quality of your recordings on the FR-AV2 than on your MixPre 3 ii ?
I can’t say I’ve noticed much difference in sound quality, but haven’t done a direct comparison. Maybe the MixPre 3 has quieter amps.
I am considering selling the MixPre 3 ii though for the following reasons:
1. Size and convenience. I can go into the field with mics plugged in directly into the AV2 and use as a ‘portable all in one’ recorder, and add cables and better mics if I need.
2. Battery life on the AV2 is much better.
3. The AV2 is much simpler to use. The learning curve of the MixPre 3 is steep and I find myself having to revisit the instructions each time I use it.
4. I prefer the files directly off the AV2.
5. The 2.5v PIP on the MixPre 3 has always irritated me.
These reasons could apply to the Zoom F3 which I have never used.
The only thing stopping me from selling the MixPre is that the screen is easier to use, the mounting points top and bottom is useful, and the fact that I paid to have the additional two channels and have not used them yet. The main reason though is that I think it’s a beautiful bit of kit and I don’t need the money!!
Thanks so much for that info. I’m asking because although I really like my FR-AV2, I have this gnawing suspicion that the MixPre 3ii preamps are just that much better. Currently there is a used one for sale nearby and I don’t know if the difference in record quality will be worth it. And I agree with you that it’s a beautiful machine but complicated.
Thanks again. You’ve given me some perspective on what may just be my lust for nice gear.
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Finally managed to acquire some IK Multimedia iRig XY Microphones. This might just be my perfect portable field recorder. They just fit into the Tascam FR-AV2 in an XY position. This is useful in the AV2 because the XLR input on the right is for the left channel.
I look forward to doing some comparisons with my Sony D100 and other recorders and microphones.
How are those mics sounding? Comparable to the Sony D100 mics?
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How are those mics sounding? Comparable to the Sony D100 mics?
Comparable to the Sony D50 I would say.
The Sony D100 remains the best sounding portable all in one recorder imho. It is crazy prone to wind noise though, and no wind protection I have tried has solved that, so the D100 is only useful in still conditions outdoors.
Whilst the iRig microphones are also somewhat affected by wind noise, I can manage it more easily. The noise floor seems to be better than the specs suggest, and as I say comparable with the D50 mics.
I am tending to use the AV2 with the iRig mics more than the D100 for the following reasons:
1. Wind noise. If it is perfectly still then I will still prefer the D100.
2. Size. The AV2 is smaller.
3. 32 bit.
4. Flexibility. I can use the AV2 with the iRig mics as a portable recorder, and then upgrade it very easily by adding my Sennheiser mics if the situation arrives.
Having said that my whole stealth field recording world has been turned upside down by the acquisition of a Zoom M4 MicTrak which seems to be totally immune to wind noise.