A handful of questions from a beginner


(newbie44) #1

Hello again,

I know it’s not nice to ask several questions at once but if create one topic per question your forum will be overloaded.

First, let me talk about myself :

Windows 7 64 bits
Ardour 32 bits (because most of my plugins are 32 bits)
Keyboard midi controller USB plugged (midi out from keyboard to computer only)
Guitar plugged in external soundcard that’s USB plugged into computer
Driver Asio4All

I think I didn’t forget anything.

Here my questions :

  • Is it possible to set Ardour to 16 bits integer by default instead of 32 bits floating?
    I know it can be changed in the session propriety menu but I forget it every time.

  • Why is the volume meter in the red so often? Even when the track are not loud or clipped? Should I worry or just ignore it?
    I put the volume on 0DB in the master and -3db max in the tracks. Is it correct?

  • Sometimes the button “feedback” turns red and the master output seems1 connected to 1 or several tracks instead of the audio out (no feedback in the sound). I reconnect to audio out and everything gets back to normal but I don’t understand why it changed in the first place.

  • Is it possible to launch Ardour without clicking on the “start”? Since I use Asio drivers any sound but Ardour is muted, which is a problem if I want to listen to a youtube tutorial and having Ardour open at the same time.

  • What’s the “Jack Transport/Time settings”? It seems that Jack is a plugin for Linux. Since I have windows, I can untick the “Ardour is jack time master”, right?

  • I won’t use the video meny, can I completly disable anything video-related and make sure that Ardour won’t search for “Video-monitor ‘xjadeo’” or stuff at the startup?

  • the “use buffered i/o” option don’t seem to change anything. Should I use it or not?


(Robin Gareus) #2

Session > Properties > Misc > “Use these settings as default” may do the trick.

but there was a bug in the past that loading another session that doesn’t use 16bit resets this. I don’t remember if this is still an issue.

There is no simple way to answer this.

When tracking I’d usually record much lower, -18dBFS or so, using float to have plenty of overhead. Also most A/D converters work best in that range (check your soundcard’s manual). Note that Ardour’s master-meter is a K-20 loudness meter by default.

Ardour can normalize at export, so I’d not worry about medium specifics (0dBFS) on the master-bus.
see http://manual.ardour.org/meters/

That is odd. It should only happen when you manually make some connections.

Preferences > General > Engine > Try auto-launch audio/midi engine

It’s not Linux only. JACK allows for inter-application routing (like re-wire). but you can safely ignore this setting.

No, not at this point in time. But it’s a good idea to add this feature. We should add a preference.

If you don’t need it, don’t use it. It’s useful for some USB soundcards and also some M-Audio ASIO drivers require an extra buffer to work reliably. Without it the soundcard’s driver directly calls into Ardour (more efficient, lower systemic latency).


(Paul Davis) #3

One answer here is that the master bus uses a K-Meter, not a peak meter. You can read about the differences (and get some links to more “expansive” reading about metering in the manual

I think you’re asking f you can use Ardour and get sound from both Ardour and also (e.g.) your web browser. Whether or not more than 1 application can use your soundcard depends on the ASIO driver for your soundcard. Many/Most ASIO drivers do not support this. Using MME will allow it (I believe) but generally requires larger latency settings.


(Chris) #4

You probably do not really want to do that. Why do you think you want to set that?

The most likely explanation is you have your record levels set too high.
This is the best place to start (articles by Bob Katz, the “K” in K-20 meter):
Level Practices part 1
Level Practices part 2

Almost certainly not. Channel and bus gain should be determined by the level of what is recorded, how many tracks are being mixed together, type or style of recording you are attempting (e.g. very clear and open classical or jazz style, vs. very compressed rock or EDM style), recorded level of the individual tracks. There is no way to give one number with any meaning of “correct.”
Start with getting the level correct for the individual track recordings. That includes the guitar interface, for external instruments you have to make sure not to set the gain too high such that the preamp distorts (unintentionally I mean, of course you can distort your guitar amp if that is the sound you want, but stay away from distortion at the sound card, that sounds horrible). If the gain is too low the noise could be too high, but that is usually less of a problem than setting the gain too high and distorting your recording, so err in the direction of a little lower than optimal.
Depending on how many tracks you are mixing, how many tracks are active at once, the levels of the individual tracks, etc. you may need track gain settings at anywhere from +12dB to -18dB. The level settings may not stay constant, many styles of music or sound effects, spoken word may need levels adjusted over time for a particular effect or for the best balance of blending vs. highlighting particular parts.

Once you have the audio recorded as floating point there is so much mathematical headroom you can recover from most mistakes you make in levels or gain settings in software. With integer audio format you have to worry about clipping the maximum and minimum levels and getting bad distortion same as with analog audio, which is why I question why you think you want integer formats instead of floating point. I think the integer format setting will just apply to the audio files format, and Ardour will still be processing internally as floating point, so probably will not make much practical difference for you, but the fact you even bring it up when you have labeled yourself a beginner make me think you probably half understood something you heard somewhere else and are incorrectly applying it to Ardour. I would suggest leaving the default settings there and just let Ardour deal with the files until you are ready to export your project to whatever format you need. If you want for example CD compatible 16 bit file format Ardour will export that once your project is finished, there is no need to record in that format.


(Mikael Hartzell) #5

I routinely set all audio track levels to -12 dBFS before I start to record audio into Ardour. I leave the master track to 0 dB.

I record tracks in one by one. I use 24 bit depth and usually have about 10 mono tracks and 3 stereo tracks in my sessions (distorted guitars, bass, vocals, drums). Setting audio track faders to -12 dB gives a sane starting point so that the mixed signal wont overshoot too easily (go to the red). You can always “normalize” audio when exporting the final mix, this means that Ardour finds the loudest part of the mix and adjusts the mix volume as loud as it can without the loudest part “going into red”.

Bit depth of 24 can store the loudest and softest sounds one can ever make and still leaves plenty of room to adjust the signal level up or down without losing any detail of the sound.


(newbie44) #6

That’s a lot of answers, thank you !

I use asio4all driver. There’s no real drivers for my soundcards. Old material.

After asking this question, I realized it’s possible to start/stop through the “audio setup” in the “window” menu. So I’ll do that. Thank you.

My external soundcard is limited to 44,1 khz 16 bits. At least, the manual says that the USB-audio-codecs properties must be set on “CD Quality, 44,1 khz 16 bits 2 channels”. I had some problems before with other VST hosts, so now I prefer to set everything on 44.1 khz 16 bits, and that’s it.

About the whole 0 db - 3db things.
From my understanding, 0db meant same level in output as in input; and anything above 0 creates clipping.
And I read (or maybe misread, I don’t know) somewhere that it was necessary to keep a 3db room before the master. So I set -3db as maximum for one track and everything else is below (-6, -5… it’s not very important for the moment as I’m just testing). And I was surprised that everything turns red so fast because I don’t really hear any clipping.


(Chris) #7

OK, I had misinterpreted your question to be about recording file formats, it is actually about output formats. Just make sure you do not get confused in the Ardour settings, you want to make sure you set that in the output device setup, not in the recording file format setup, the two settings are different and unrelated.

You are mixing up gain and levels. In the context of gain, 0dB means the output is the same as input, whatever that level may be. In the context of level, 0dB means the signal level is the same as the reference level. There are subtleties there in context, digital signals are often referenced to full scale, usually notated dB FS, in which case 0dB means the signal is at full scale, i.e. cannot be higher without clipping. In other contexts, for example analog signals, 0dB refers to the nominal average level, so it is common for example to see analog interfaces which reference 0dB nominal, and anywhere from +18dB all the way up to +24dB maximum signal level accommodated.

That is a good rule of thumb for analog interfaces. Internal to Ardour all the calculations are handled as floating point values, so as long as the signal at the output of the master bus to your audio interface does not go above 0dB FS you should be OK.

As I pointed out before, when you add multiple signals together the levels increase, depending on how similar the material is could be anywhere from 1dB to 6dB when two tracks are mixed together. It does not take many tracks mixed together before the master bus level is possibly up to 10 dB higher than any single track.

You also have to be aware of where you are metering. You can check signal levels at the input to a track, output of a track, input of a bus, or output of a bus. You can go above 0 dB on internal paths, but it makes keeping track of proper levels difficult. Although not required, I find it easier to treat signal paths in ardour similar to how I would in an analog mixer, try not to go above max level even on internal paths, although if you do the audio will not actually clip as long as you bring down the levels before sending to your audio interface.


(newbie44) #8

Thanks, I think that I’ll do the same.

I have a question. It may be a silly question but… don’t we lose some details when recording at low volume ?
I mean, if you don’t hear the harmonics and background sound on the monitors headphones are they still recorded on the track?
And you can retrieve them during the mixing process (with gain, level, eq, compression and so on…)?


(Mikael Hartzell) #9

Setting the audio track level in Ardour only affects Ardour output volume, not the recording. You adjust the recording level with your audio cards volume knob. Set it as high as you can so that it will never clip while you record. How high you can get depends on what you are recording. Its better to lose a bit volume when recording than lose the whole thing because of ugly clipping noises.

If it is hard to guess how loud the performer will be during a performance it might be good to record him to two (or more) tracks one with much lower volume than the other so that you have some kind of recording if worst happens.

If it is just you playing an instrument at home then set the volume on the audio card high and if it clips, just turn it down a bit and record again.

Yes most of what you hear in the room while recording a microphone will be recoded also. That’s why the microphone usually is placed very close to the sound source, and as far away from the noise source as possible. This will make the sound we want much louder in the recording than the sound we don’t want (computer noise, sound reflections from the walls, etc). Then you have a big difference between the “good signal” and the bad (noise) and the noise can’t be heard when creating the mix. It is true that if there is lots of noise in the recording using a compressor will probably bring it up.


(newbie44) #10

Thank you.

Set it as high as you can so that it will never clip while you record. How high you can get depends on what you are recording.

For the moment, it’s mostly virtual instruments.

My MIDI keyboard is plugged to my computer and set to use VSThost (of Hermann Seib).
All the VST synths, amps, effects, drum machine, etc. are loaded in VSThost and from there, the audio data is routed to Ardour via Jack.

I’ve hesitated between sending midi data or audio data to Ardour but for the moment I prefer to work on audio.

(I also have a guitar but I plug the jack into an external soundcard and the soundcard to the computer. I don’t think I’ll use real microphones).

Set it as high as you can so that it will never clip while you record.

I heard it was important to keep the input gain around -18 dbfs and make sure the peaks don’t exceed -6dbfs. I can achieve that with the master volumes inside VSThosts but… it bothers me a little because it doesn’t sound the same.

For instance, I use a DR-910 Vst drum machine. I select one the preset, I like how it sounds, I don’t want to change anything.
I route the whole thing to Ardour, I set the meter on “input” and make a test : it’s red. So I lower the master volume in DR-910 until the meter peaks at -6dbfs and about -18 on average.

All is good, except that what I hear in my headphones is much lower and I have less detail, I can barely hear the kick or the toms… You see?

My first idea was to make what I hear in Vsthost --> jack --> ardour --> my computer speakers; as close as possible as what I hear in Vsthost --> my computer speakers. But I see now that it’s not possible.

Good idea. I’ll do that.

Jeez! I’m asking all these questions and I’m not even a real musician. I’m just doing it for the fun. :smile:


(Mikael Hartzell) #11

The phenomena that you describe where an instrument sounds different when played at different volume levels is caused by our ears. Its a well known phenomena. All the frequencies are there in both the loud and softly played version, but the ears sensitivity to low and high frequencies will be worse at lower volumes. When the volume increases our ears will hear more of the bass and high frequency sounds. This is why music sounds better when played loud.

The instrument that sounds better coming directly (louder) from the vsthost than Ardour (with less volume) should sound the same if you raise the volume when listening through Ardour. So you don’t really lose fidelity when recoding at less than maximum level, all the frequencies will be there.