Complete Classical Music workflow

I will do now

I tried three vias:

  • Downloaded it
  • Copy and paste the raw Data.
  • Copy and paste the first bash.

Now I am using the downloaded file.

Here is:

luis@antix1:~
$ sudo dos2unix /usr/local/bin/album-ebur128
dos2unix: convirtiendo archivo /usr/local/bin/album-ebur128 a formato Unix…
luis@antix1:~

WORKING NOW!!! :smiley:

Thank you @anon60445789!!! This is the output of all my recorded files

This is great. I do wonder where those Windows/DOS issues crept in?

Anyhow, the next step for me is trying to figure out how to get a proper integrated and loudness range reading for the whole album and not just an average(mean) value as when run as a single album WAV, the values are not right. A good first step though. @Aleph which Linux OS are you running?

Yes! Good luck with the script, I’m sure you’ll find a way.

Now go enjoy with your family, please :wink:

And thank you again!

Antix19:

inxi -F
System:
Host: antix1 Kernel: 4.20.0-16.2-liquorix-amd64 x86_64 bits: 64
Desktop: IceWM 1.5.5+git20190610
Distro: antiX-19_x64-full Marielle Franco 16 October 2019
Machine:
Type: Laptop System: Dell product: Latitude E6430 v: 01
serial:
Mobo: Dell model: 0CPWYR v: A00 serial: BIOS: Dell v: A22
date: 02/21/2018
Battery:
ID-1: BAT0 charge: 49.1 Wh condition: 49.1/57.7 Wh (85%)
CPU:
Topology: Dual Core model: Intel Core i7-3540M bits: 64 type: MT MCP
L2 cache: 4096 KiB
Speed: 1972 MHz min/max: 1200/3001 MHz Core speeds (MHz): 1: 1276 2: 1246
3: 1271 4: 1440
Graphics:
Device-1: Intel 3rd Gen Core processor Graphics driver: i915 v: kernel
Display: x11 server: X.Org 1.20.4 driver: modesetting unloaded: fbdev,vesa
resolution: 1366x768~60Hz
OpenGL: renderer: Mesa DRI Intel Ivybridge Mobile v: 4.2 Mesa 18.3.6
Audio:
Device-1: Intel 7 Series/C216 Family High Definition Audio
driver: snd_hda_intel
Sound Server: ALSA v: k4.20.0-16.2-liquorix-amd64
Network:
Device-1: Intel 82579LM Gigabit Network driver: e1000e
IF: eth0 state: down mac: b8:ca:3a:d7:09:24
Device-2: Broadcom Limited BCM4313 802.11bgn Wireless Network Adapter
driver: wl
IF: wlan0 state: up mac: f4:b7:e2:c7:cc:1b
Drives:
Local Storage: total: 298.09 GiB used: 7.86 GiB (2.6%)
ID-1: /dev/sda vendor: Western Digital model: WD3200BPVT-75JJ5T0
size: 298.09 GiB
Partition:
ID-1: / size: 57.42 GiB used: 7.86 GiB (13.7%) fs: ext4 dev: /dev/sda7
ID-2: swap-1 size: 7.96 GiB used: 0 KiB (0.0%) fs: swap dev: /dev/sda5
Sensors:
System Temperatures: cpu: 47.0 C mobo: 38.0 C sodimm: 38.0 C
Fan Speeds (RPM): cpu: 2711
Info:
Processes: 169 Uptime: 1h 51m Memory: 3.76 GiB used: 1.35 GiB (35.8%)
Shell: bash inxi: 3.0.36

OK, will do. But first, I switched over to antiX 17 and downloaded script from pastebin as if I was another user and I realized that pastebin is adding Windows CRLF line endings which is not cool :frowning: Also had to use dos2unix to fix… I think once I get the code where I want it, I’ll just host it on Google Drive or similar.

Likely too late now. While I appreciate your enthusiasm, this is now getting seriously off-topic.

I wish moderation had stopped this and suggested a new thread when the “listen to my recording” spin started two weeks ago. I very much liked the beginning of the thread which was actually about describing the workflow.

@anon60445789 It looks like you’ve read most of the replies since. Has any useful information or hints come up since that you could added to the original post at the very top? That would be helpful for future visitors. Thanks!

I’m sorry Robin, you’re right, it was my fault.
The subject has been deriving around my project…
Better leave this thread.

I’m not the type of person to shut someone down when they are genuinely seeking feedback even though the conversation wasn’t perhaps sticking to my original idea. Honestly it seemed like it might be the perfect way to document a thorough workflow from start to finish with @Aleph’s recording project . Given there is no option here for private messaging I personally felt this might work. I agree that, despite my efforts to engage with other aspects (note my questions that appear at regular intervals) there were no real takers. And I want to make something crystal clear here: @Aleph is in no way to blame. I and others engaged with his recordings in meaningful ways and I think, in many ways, this has been useful to those who are/were taking part.

The topic is “Complete Classical Music Workflow” so much of this, including calculating loudness on the command line seemed appropriate to me (and answered based on a direct question from the community). I think it was clear from the beginning that Ardour would play just a partial, yet important role in discussions. However, I do take your point. For a new user reading through this conversation it might feel overwhelming.

@x42 You are correct. I’ve read all the replies and spent hours researching and typing back kind, helpful responses. Again, if I came up with a script (like my DDP one) that I thought might be helpful to the community and someone was having issues, how else to help them?

It is sad, but I’ll be joining you. “Moderating” is about removing people if they use hateful language or profanities, none of which have occurred here. I’ve enjoying seeing the conversation morph / take shape based on the needs of the users/community.

Again, I hope you think this has been helpful and I’m truly sorry if you’ve been made to feel like this is your fault. It isn’t and I, for one, have enjoyed it. I genuinely set up the conversation to start a useful discussion and, most importantly, help people. I feel deflated, honestly.

As the moderator in question here I’ll chime in:

Moderation as x42 mentioned wasn’t about removing people from the topic, it was about making sure the topic matched the conversation. This can be about helping people as much as anything, as those looking for a classical music workflow discussion might not be as interested in the technical details and issues of the script here and the technical support, and similarly those looking for similar assistance to that might not find it in the classical music workflow topic.

I will disagree with x42 only on the timing needed, I don’t have a problem so much with the discussion around the specific recordings as they were discussing something related to the primary topic, but he is correct in that the troubleshooting of the script and related discussion should be in a different topic is all.

There is no need or desire to shut down the conversation, or remove people from it, just a desire to make it clearly labeled and pertaining to the topic at hand. There is great discussion here that we want many people to find, that is all.

If there had been anything that warranted a removal from the conversation I would have stepped in much earlier and much stronger.

     Seablade

What @seablade said. Moderation sometimes also involves moving a discussion in the right direction.

Instead of accumulating replies, I was thinking of just opening a new topic in made-with-ardour category, where mixes are usually discussed, or the how-do-i category for the bash-fu. – discourse can automatically cross-link when a topic is mentioned. – Maybe consider that for the next time. Thanks!

That being said, please do carry on! There is some excellent information hidden in this overall discussion, and thanks for starting this thread in the first place.

Yes, concatenate the files without gap and pipe it to ebur128. For example with sox:

sox file1.wav file2.wav file3.mp3 -t wav - | ebur128 --lufs -
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Thanks, I was in touch with @Aleph and we were thinking of starting a conversation purely on his flute project. I’m happy to continue the classical workflow thread too but how to proceed? Perhaps a soft reset and starting a discussion on each of the major sections I outlined in the OP?

How about just keep going here? In this thread the ship has already sailed. – It was mostly the off-topic terminal debugging that triggered me.

If you wish we can delete the last couple of posts pertaining to the “moderation” discussion. There’s no need to bug posteriority with it.
@seablade any opinion?

Here we go:

Let’s take care of any issues/feedback over there and continue general classical workflow here. Thanks @x42!

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I think that fits very well thank you.

In as far as the discussion about the recording in particular, here is how I would think about it. If the discussion is about the recording technique (Or workflow) for classical flute, I would think here would be fine, though you could start a new thread if desired. If the discussion is about that particular recording and/or song, then lets start a new thread about it in the ‘Made with Ardour’ category.

In as far as the moderation discussion, lets keep it present long enough to at least confirm all parties have read and understood it. If there are any other questions, email me (gmail: seablaede note the extra ‘e’) and I will be happy to take it up there, but I don’t feel like there is a need to continue it here and so long as no one objects I may hide those posts here before long.

      Seablade
1 Like

So I think I can keep asking general questions about the classical music workflow here and I will start a new thread when I want to ask specific questions about my project.

Thank you friends.

I thought that we could go through each of the sections I outlined in the OP and open up for further discussion. Perhaps we could first discuss capture of audio via audio interface and microphones?

I would be interested to know about what levels people set when recording in 24-bit, whether people are now using the new 32-bit float field recorders (MixPre ii and Zoom F6) and what people might recommend for a linux-friendly audio interfaces and good stereo pairs specifically for classical capture where preamp quality is paramount. For what it’s worth, I just picked up an Audient iD44 (tested in Windows but not in Linux as yet) to add to my MixPre6 ii field recorder (audio from SD card into Ardour). For microphones I just purchased a pair of Shure KSM141 to supplement the AT4022/4021, sE4400/sE8 and Behringer B5/B2 Pro pairs I own. I generally use a stereo array such as ORTF or spaced pair if using omni (occasionally blending two arrays when there is limited sound check). I’m experimenting with NOS, DIN and EBS based on the “sweet” spot in the room, distance from ensemble and spread of musicians. Much of this you already knew. What about you?

1 Like

Great topic, and I appreciate the continuing conversation on these points. When I was using Linux, my trusty Sound Devices USBPre 2 interface did a great job: its preamps are similar to those of the legendary 788t and it’s a very versatile little device. It can be USB-powered or plugged into a power source; it can also be used as a standalone preamp not connected to a computer. But I did find myself yearning for more inputs and ultimately went for the Mixpre 6, first generation. For a very high-quality two-input preamp today I would probably get this: https://www.sonosax.ch/product/sx-m2d2/. It’s class-2 compliant and runs on Linux.

As discussed previously I like using a field recorder, in part because there’s no fan and it’s a simple self-contained unit. A couple of weekends ago I ran into a problem with my MixPre (my fault, not the recorder’s) so I switched to recording through my USBPre 2 into Mixbus on a Thinkpad (running Windows 10) and recorded for several hours without a problem and without the fan ever kicking in. It was nice to have all that screen real estate compared with the MixPre’s tiny interface, so I’m tempted now to start using the laptop. But if I start doing this more regularly I’d switch it over to Linux, which I trust more than Windows for critical applications.

My recording gear is ancient, and whenever I think about upgrade, SD USBPre2 first comes to mind :smiley:. But, it is very expensive in Europe… In that class (1k-1.5k€) there are other good options: Digigram Cancun, Merging Anubis, Sonosax (thanks, @bradhurley) … Then there is option to buy preamps and converters separately - Mytek ADDA 8x192 with optional usb card is class compliant, D.A.V. and Line Audio both make fine inexpensive preamps. These are all costly options. The question is how bad are inexpensive audio interfaces these days. Audient, Presonus, Focusrite, even Behringer (via Midas) all have know-how to produce top class preamps. The question is only how much compromises they made to lower the price, and how much do those compromises affect sound quality. Do you remember that Gearsluts thread about people having to decide which is which between a Grace preamp and ART Tube MP?