justinpbrown71

20 hours ago, Cactus Music said:

This might be similar to an issue I have using a Motu M4.  I’ll get distorted recordings. Not play back ever but sometimes new audio recording is distorted. 
It only happens when there’s been outside activity like I was using Sony Vegas or You Tube. 
The only way to get back audio is a total reboot of computer and interface.  
So next time it happens try that. 
 

One other place to look for audio issues is matching sample rates. I explain how to set this in Tutorial #1. 

 

Thanks for your reply, John.

I have watched your recommended video a couple of times and set up everything as it should be. The sample rates match between the hardware, Windows and CW.

I have reduced the buffer size to 1024 for recording and playback, but still I get the random 'robotic/frequency modulator' noise effect occurring during record and playback. Sometimes I get 15 minutes of recording time before it starts. Sometimes only on the track I'm recording onto, though usually after a short while it  appears on all the other tracks playing back.

I don't think it's a power issue; I've been monitoring the power draw and it's not too high.

I've stopped some unnecessary background services and have no AV running apart from Windows native one and Defender. Updates are halted.

Here are my current settings:

 

I have an acoustic guitar, with pick-up, inputted through a Zoom R8 interface with the latest Zoom R8 ASIO driver installed (Windows 10).

The problem I have is that after about 1 minute of recording, the guitar signal suddenly changes to sound like it's being put through a frequency modulator effect, and the recording played back is the same.

I also get this when playing back previously recorded tracks (not recorded through the interface), but only when using the Zoom R8 interface, and intermittently.

I've read that this could be because of excessive cpu use due to the latency being too low, but my cpu use is not increasing above about 30% (RAM abt 60%). I have tried changing the buffer size from 256 to 512 and 1024, both in the Zoom ASIO control panel and Sync and Caching > File System, but there's no improvement.

Here is an image of my Driver Settings in CW with the buffer size set at 256:

UPDATE - Now the effect is currently permanent on all the tracks of one project, but if I export and play them back in WMP (via the interface) they are okay. Other projects are playing back okay.

UPDATE #2 - Having exported all the individual tracks and testing them (they're okay), I returned to CW, and without re-opening it played the original project which all of a sudden played back perfectly through the interface. I had made no changes to settings or anything at all.

UPDATE #3 - I have tried recording on other projects, as I thought the issue could be due to using specific VSTs, but even recording the guitar clean, with no VSTs previously added to the track, after a short while the sound of the guitar becomes distorted.

Ah, three weeks later I discovered the issue. Very simple...

Windows recognises the Zoom R8 interface as a microphone.

In Windows privacy settings I had access to the microphone disallowed, as well as access to it by desktop apps.

I also discovered that Windows was automatically replacing the native Zoom R8 drivers with Microsoft generic audio drivers, but was able to fix this issue by unhiding the list of (what Windows considers) non-compatible devices during the manual driver installation process, and select the Zoom R8 device.

18 minutes ago, scook said:

install the driver and update the firmware (if necessary) https://zoomcorp.com/en/us/digital-mixer-multi-track-recorders/multi-track-recorders/r8/r8-support/

 

It is very common for ASIO drivers to work at a single bit-depth, usually 24 bit.

I installed the latest R8 driver yesterday. I'll check the firmware...

UPDATE: The firmware needs updating. Currently I have the previous version. The R8 doesn't detect the firmware file on the SD card though. I've followed the instructions correctly 3 times. I'll have to get in touch with Zoom about this.

2 hours ago, Cactus Music said:

Follow the instructions in this video and you will have sound. I guarantee it. 
 

 

Hi John,

Thanks for video tutorial.

I have followed the instructions, step by step. I had a positive sound test with the Windows test, but:

1. The Zoom R8 audio interface is not showing in the Audi Devices, nor Driver Settings lists in CW when I select ASIO, but is when I select WASAPI SHARED.

2. When ASIO is set, the Audio Driver Bit Depth is greyed out at 16, not 24, as is set in Windows.

3. The 64-bit Double Precision Engine box is pre checked, unlike in your video.

4. The buffer size slider is set to FAST and greyed out. The ASIO reported latencies are all set at zero.

5. Sync and Caching: there is no device showing; the dropdown menu is empty, and the Use ASIO Reported Latency checkbox is unchecked. It's value is set to 1234.

If I set Playback and Recording to WASAPI Shared I can see the Zoom R8 in all the places it seems it should be when this setting is set to ASIO in your video. Set to WASAPI Shared I get playback sound, but no audio input from the guitar or built-in mics from the Zoom R8 is registering in CW.

Finally, I have to disable the NVIDIA sound device in Windows to have the Zoom R8 'stick' as an option in the Sync and Caching device dropdown (when set to WASAPI Shared).

Last year when I used the Zoom R8 as an interface to record in this manner I didn't have to keep entering into the Windows settings and changing them every time I wanted to use the R8, and when I had finished using it, to turn on my monitor speakers. Is there no other way than having to enter the Sound Settings in Windows and having to change them every time I want to record?

Am I missing something, or have I misinterpreted your instructions?

I have connected the R8 to the pc to record guitar directly to Cakewalk. I have the R8 side sorted, with an input signal, after following the instructions in the R8 manual. I have installed the latest R8 driver. The R8 is recognised in the Cakewalk preferences, in all instances except for the Audio Playback and Recording Driver Mode, and under the MIDI options for Controller/Surface, where there is nothing listed.

I have tried the Asio and Wasapi Shared options so far. I had ASIO4ALL installed, but when I selected ASIO option CW popped up a window stating that this driver is not compatible. I read elsewhere that the ASIO4ALL 'wrapper' can 'hijack' other ASIO drivers (I'm not 100% sure what this alludes to). So, I uninstalled ASIO4ALL and rebooted. Now in the Audio Playback and Recording Driver Mode, when I select ASIO I do not see an error window. I assume then that now the R8 driver is recognised and accepted.

Still there is no audio signal registering in Cakewalk.

When I arm an audio track and then select the record button to begin recording, I get this message:

'Unable to open audio record device. Device may not support the current project's audio format or may be in use.'

After creating a new MIDI track, manipulation of the R8 controls, triggers a flash of green at the top of both the track and buss level meters, but I get no signal from my guitar.

I can hear all the project tracks playing within CW through the headphones plugged into the R8.

Anyone can help? Many thanks.

On 7/22/2022 at 4:59 PM, Cactus Music said:

FYI my videos are short and to the point. I don’t even have intros and I don’t beg for subscriptions. That would waist  time. People just need the facts. I made them per topic.

If you do watch the ones I listed you will be miles ahead in understanding how to produce a top quality master. There’s much more to understand about Mastering  than what I cover, but for most new comers this is a great starting point.  I can totally understand not wanting to waist time on yet more videos but mine are different. I only made them because I got tired of typing the same answers over and over in this forum  

So the sort answer is it seems you have not set your Master bus section up properly.  Only the Master needs to be at Unity.  It's all about understanding Cakewalks signal flow. All this is covered in detail in the ones you didn't watch. The Video's about Exporting is NOT about mastering. It assumes you've got that under control. 

You shouldn't be messing with Windows volume setting either, unless you don't have an audio interface.  We normally adjust monitoring levels with Hardware. You also might benefit from Tutorial # 1 about setting up your audio system correctly. 

I've just got back to my musical project and wanted to thank you for the video suggestions, and indeed, the videos themselves.

Please don't take me the wrong way, I do appreciate that you've given your time and expertise into creating the comprehensive set of videos. I have definitely learned much from watching them. My initial remarks were merely expressing that I find it much easier to learn from reading text than watching videos. It's just a personal thing.

Anyway, you have my gratitude for your assistance.

19 hours ago, John Vere said:

All volume settings need to be at unity if that is the source for your export. I have tutorials on exporting and mastering projects that will clarify what you are trying to do. 
 

#15, 22, 24 and 28,29. 
 

 

Thanks very much for the tutorial links, John. You have provided links to 6 videos covering mixing, mastering and exporting, over 45 minutes. Where amongst this information is the solution to my issue?

Is the tutorial information available as text? I find it very difficult to absorb information from videos, and have to keep replaying them over and over and eventually converting them into document format, otherwise I forget the information.

"All volume settings need to be at unity if that is the source for your export." - How can unity be the source of export? I don't understand this statement.

I have made many recordings in the past using CW, but not for a long time. This is the first time I have had this issue, so I expect there is a setting/s I have applied incorrectly, or not at all.

Many thanks.

[I have watched the two videos about exporting, and am already exporting in the manner explained. There is no troubleshooting covered in the videos. I have also imported an exported song back into CW and it plays back at a good volume. So, I don't know why I have to push VLC to maximum volume and increase the headphones from 50% top 100% volume and still not get the exported song to play back at an equivalent volume level.]

I have just returned to using Cakewalk and am reacquainting myself with what I used to know. My experience level is low.

I have an issue with the playback volume of songs being very low after export, perhaps about 50%.

In CW the levels seem to be fine, with the meters set to monitor RMS & Peak, and the headphone level set at 50% in Windows.

Export settings I have tried are with Entire Mix through the master bus or the headphones as source.

4 hours ago, Andres Medina said:

"Also, is it possible to copy pan settings to multiple tracks, or pan them simultaneously, as I will be adding these to 34 tracks, 17 L and 17 R?"

--- 

Yes, select all the tracks you intend to work on; Ctrl Click while you pan one of them. All tracks will follow.

Another method: select all tracks - Assign them to a group.

See videos.

Great! This method is ideal.

Thanks very much and happy new year!!!

4 hours ago, bdickens said:

Put the compressor on a bus and route the tracks to it.

Thanks very much.

So, if I place fx on a bus, (instead of on the clip itself), then they always affect the clip at the output stage, after any fx I might add to the clip directly?

If so, then this is probably the best way to apply compression?

I'd be grateful for any tips.

I want to add compression to 21 individual clips on 21 tracks simultaneously. The settings for the compression will be the same.
How can I do this?
If it is not possible, can I add the compression fx to one track and then copy and paste it to the others?

Also, is it possible to copy pan settings to multiple tracks, or pan them simultaneously, as I will be adding these to 34 tracks, 17 L and 17 R?

Thanks.

On 9/28/2020 at 11:53 PM, Will_Kaydo said:

Its highly Audible on my phone too. Sounds like no Pop filter? 

It was a problem with the speakers.

Speaker problem.

5 minutes ago, fossile said:

getting a lot of artifacts - maybe somewhat due to compression but it seems more like edges of audio were clipped without cross-fades or simply silenced on non-zero passing. also, no much high freq so it's highly compress mid-range which impacts clarity of bit as well. could be mp3 format killing some highs.

here's the waveform via RX-7

The recording I attached is not of the original recording. I posted it only as an auditory example of the popping effect through the speakers. I have no idea how it was recorded. There are probably all sorts of background noises present.

I am recording an audio book. I have an issue with playback, popping plosives. It only occurs when played back through my friend's computer speakers, which have no issues playing back other recordings. The recording plays back fine through my and his headphones, and also through my and another friend's laptops.

My guess is it's a compression issue. These are my settings using ReaComp:

Threshold: -30
Pre-comp: 0
Attack: 3ms
Release: 100ms
Classic Attack: ON
Auto Release: OFF
Ratio: 4:1
Knee: 3db
Low/High Passes: Default
RMS: 5ms (default)
Wet: 0
Dry: -20
Auto Make-Up: ON or OFF
Limit Output: ON

My chain consists of ReaEQ, set with standard low and high pass filtering, followed by the above compression and then another, flat ReaEQ with a gain boost of around 2 to 3db.

My recording volume is -5db. I have been singing with mics for about 30 years and so have good proximity control. My mic is a condenser with wind shield, set up approximately 30cm away from me, in a sound absorbent recess.

The recording has been tested by the ACX analyser and is well within the required parameters for peak, RMS and noise floor.

I attach a recording of the playback through my friend's computer speakers where the popping sound can easily be heard.

Thanks.

popping.mp3

1 hour ago, John Bradley said:

Is the particular room sound important?

If not, I'd think the easiest thing to do might be to gate and compress the audio, edit out whatever unwanted noises made it past the gate, and then add some synthetic ambience (say, a nice covolution reverb) afterwards.

Just a thought.

Though fwiw, those recorded signals look very low, and might be getting somewhat lost in the room sound even while you're talking. More sophisticated repair might be necessary. (e.g. RX6/7/8 have de-noiser and de-reverb plugins, I believe.)

The room tone is the ambience of the room I record in, so is necessary to use in conjunction with the recordings I am doing, for consistency. The room tone is simply a recording of the ambience of the room without myself in it. It is a specific tool for patching inbetween phrases where lip smacks or other undesirable sounds might occur during recording.

To the narration tracks I have added EQ, compression and boost, to achieve the specific volume parameters required by ACX for audio book use, for noise floor, peak and RMS. I have added the exact EQ, comp and boost to the room tone track, so they are equal in volume and noise level.

The signal looks low because it needs to be; between -18 and -23 db RMS, peaking at -3db, with a noise floor of at least -60db. These are the ACX requirements for audio book submissions.

45 minutes ago, scook said:

I see T1 is muted.

When comping the lanes should not be muted, the clip(s) need to be muted.

Yes, I was trying to test the difference in playback when each of the take lanes was muted. I am so frustrated with this work. There is only problem after problem. I don't know what to do. I will have to find another method where the volume descrepancy doesn't occur. Thanks for your time and input.

2 minutes ago, scook said:

The take lane track is the original track. I scaled the track to make it easier to see. There is no volume difference between the parent track and take 1.

 

You can see in my screenshot, Track 4; the take track copy of the original track is visibly larger then the original representation above it, which, I believe, signifies an increase in output volume. I have not increased this myself. How else can the increase in original track volume, (or decrease in tone room track volume), be accounted for?

On 9/18/2020 at 4:40 PM, scook said:

 

I have a problem with this method. As you can see from your video, the volume of the take lane track is higher than the volume of the original track; how can this be rectified? How can I prevent the volume from being automatically raised? The visual representation of the the take copy of the original track is noticeably larger.

What I encounter is that the sections of room tone I add (in place of the spaces between the narration phrases) play back at a lower volume than the original spaces between the phrases on the narration track, even though outside of the take lane comping setup, the room tone track and spaces inbetween the narration phrases on the original track sound equal in volume.

Many thanks for the explanation. I appreciate your taking the time to help. I now have a different issue I have to take care of before I can embark on this task...

1 hour ago, scook said:

You actually have two takes, the narration and the room sound

Here is a small video showing how to substitute "silence" for part of a clip. It consist of the parent track, the take lane with the "room clip" and another with the actual performance. The lanes are zoomed in quite a bit but you could go further. Notice how one can progressively replace the clip with sound with using the other clip.

 

Thanks very much for the video. I will try that method.

But which am I editing? The original track or the take? If the take, then I can see (as in your video too), that the audio levels have changed, the audio looks to be louder. Also, I want to export my original track before editing, but when I delete the take, the original clip is also deleted.

I wish there was just a simple copy and paste option for this.

Also, having duplicated/copied/bounced the room tone track to a size adequate to use as a take alongside the original audio clip, when I add the room tone track to a take lane, the excess is also added onto the end of the original clip. Why is this?

9 minutes ago, scook said:

Yes, I understand.

Comping will do this by allowing you to replace one part of a clip with another

As an experiment

• create a track
• open the take lanes
• copy a narration clip into the lane
• add another lane and copy the room sound into another muting the clip
• try comping between the two

 

 

Thanks for the instructions scook, but I simply don't understand. I have read your instructions and the comping instructions on the page via the link, but cannot relate them to what I desire to achieve. I have only one take. This method seems to be for compositing multiple takes into one.

Thanks for your time anyway.

14 minutes ago, scook said:

Could add the room sound as muted clip(s) into a second take lane and use comping to replace the unwanted sounds in the narration lane with the room sound

or

use volume automation between two tracks

Sorry, I don't understand. Comping doesn't seem relevant, unless there's something I'm missing about this method, though I did read the page via the link you posted.

Also I have, essentially only one track (per chapter of narration, approximately 10mins each) which is edited for best performance. But there are lip smacks and occasionally unwanted sounds between phrases; it is these sections of unwanted sound I wish to replace with room tone.

2 minutes ago, scook said:

It is still in the thread

The default sort order of the Q&A section is by likes NOT by date. The sort option is on the right just below the question post. At least that is where the option is located in the default PC browser theme.

Thanks scook. How confusing!!! Well, it's there twice now. I haven't yet been able to figure out how to delete a post...