Autoselection of instruments and other questions from a newcomer

[mptntguru]

Greetings to everyone. I have been advised using openmpt and was amazed by the quality of sound. It is better than sound from any daws even most expesive ones. Why is it so? what's the secret.
Question 1. which algo has highest quality? polyphase or lmms?
2. From whichever column i select a cell I get the same instrument playing. WHy is it so? I have three channels assigned to 3 copies of kontakt vst plugins loaded in instruments tab as fx1, fx2, fx3. But whichever cell i choose between three columns in pattern tab, only fx1 is playing sound (or only for example fx2 depending on what i have selected in the field above.
3. Why is the maximum length of columns is 1024 cells which equals to 4 m 29 sec song? Does it mean that i can't compose songs longer?
4. Suppose there is a possibility that after 4 m 29 sec notes begin to be inputted to other pages corresponding to the row of numbers 1 2 3 4 5 ... so on. If i have a long sustained note at the end of column one will it be broken if it should continue on page 2?
5. When i record from midi keyboard sustain event cc 64 are not recorded at all. Only notes and velocities. What's wrong?

[manx]

1. which algo has highest quality? polyphase or lmms?

The polyphase filter implements basic anti-aliasing during resampling and is thus slightly higher quality, which is also why it is the default.

The fact that the XMMS-ModPlug filter is listed below in the drop-down box has mainly historic reasons.

[mptntguru]

6. A guy has told me that he does some complex routing including between channels in openmpt using some vstforx plugin. When I asked him for details I got no answers. I don't have vstforx but i have metaplugin v.2.5. Is it possible to get interchannel routing with metaplugin the same was as via vstforx? If yes than perhaps it's also possible to use one copy of kontakt for 16 instruments because i could create fx1 with kontakt vst and 16 more fxs receiving signal from fx1.
My intend is to use openmpt as a full featured daw.
7. How do i do automation in openmpt?
8. There is no render button or menu item. So how do i render the project to a song?
9. How do i process waves? I have been advised first bouncing midi tracks to wav tracks and remedy phase issues before rendering.

[mptntguru]

@manx
And what does amiga mode do? Does it process sound with amiga like saturation?

[Saga Musix]

There is no secret, and in particular OpenMPT's resampling filters are not of the highest quality - however often objective quality is not what people want.
An objective comparison between the sample players of various DAWs (including OpenMPT) can be found at http://src.infinitewave.ca/

1. which algo has highest quality? polyphase or lmms?

In addition to what manx wrote:
They have a similar quality, but one might be more suitable for some scenarios than the other. In particular, ModPlug-XMMS (not LMMS, that's yet another software ) is more flexible in that you can configure the filter bandwidth (more important) and  filter window (less important) yourself, but on the other hand it lacks the anti-aliasing capabilities as manx described.

2. From whichever column i select a cell I get the same instrument playing. WHy is it so? I have three channels assigned to 3 copies of kontakt vst plugins loaded in instruments tab as fx1, fx2, fx3. But whichever cell i choose between three columns in pattern tab, only fx1 is playing sound (or only for example fx2 depending on what i have selected in the field above.

Unlike in DAWs, there is no strict assignment between pattern channels and audio sources. You can place an instrument in any channel you like, so the second column of the entry - the instrument column - is important, because it tells you which instrument plays.

By the way, the resampling options (Polyphase and ModPlug-XMMS) do not matter at all if you use plugins like Kontakt, because they do their own resampling. This is also true for other DAWs. The audio output of a plugin is the same no matter which DAW you run in it.

3. Why is the maximum length of columns is 1024 cells which equals to 4 m 29 sec song? Does it mean that i can't compose songs longer?

The practical reason is because the file format does not allow for infinitely big patterns. However, the more user-relevant point is that you should be using more than one pattern. A single pattern can not be longer than 1024 rows, but you can have hundreds of different patterns. To keep things tidy, it is more advisable to have two to four bars per pattern (whatever you are the most comfortable with), which is typically 64-128 rows. Some people also like to have longer patterns, but it's rarely needed to have longer patterns than 1024 rows if you make use of more than one pattern.

4. Suppose there is a possibility that after 4 m 29 sec notes begin to be inputted to other pages corresponding to the row of numbers 1 2 3 4 5 ... so on. If i have a long sustained note at the end of column one will it be broken if it should continue on page 2?

Notes are sustained between patterns, there are no automatic note-offs, ever.

5. When i record from midi keyboard sustain event cc 64 are not recorded at all. Only notes and velocities. What's wrong?

Nobody has implemented that yet, that's what's wrong. OpenMPT's MIDI implementation is not very complete because it is not based on MIDI internally at all, but support for CC64 is planned to be added in one of the next OpenMPT versions.

[Saga Musix]

6. A guy has told me that he does some complex routing including between channels in openmpt using some vstforx plugin. When I asked him for details I got no answers. I don't have vstforx but i have metaplugin v.2.5. Is it possible to get interchannel routing with metaplugin the same was as via vstforx? If yes than perhaps it's also possible to use one copy of kontakt for 16 instruments because i could create fx1 with kontakt vst and 16 more fxs receiving signal from fx1.
My intend is to use openmpt as a full featured daw.

Since I have not used any of those plugins, I cannot tell you what is possible with them and what isn't. That is more a question about the features of those plugins than about OpenMPT.

7. How do i do automation in openmpt?

For example using parameter control events.

8. There is no render button or menu item. So how do i render the project to a song?

You are probably looking for the "Export as lossy/lossless" menu items in the File menu.

9. How do i process waves? I have been advised first bouncing midi tracks to wav tracks and remedy phase issues before rendering.

That is a very broad question. OpenMPT has a limited sample editor but if you want to process wave files further, it might be advisable to simply assign the instrument that is assigned to the sample to a VST effect.

And what does amiga mode do? Does it process sound with amiga like saturation?

There is no saturation, but the oversampling characteristics of the Amiga are emulated more closely than with the other filters. In particular, the Amiga has practically no aliasing, which is something this filter emulates.

[manx]

And what does amiga mode do? Does it process sound with amiga like saturation?

There is no saturation, but the oversampling characteristics of the Amiga are emulated more closely than with the other filters. In particular, the Amiga has practically no aliasing, which is something this filter emulates.

The Amiga resampler will not be applied to all modules even if enabled, though. See the paragraph in the manual: https://wiki.openmpt.org/Manual:_Setup/Mixer#Use_Amiga_resampler_for_Amiga_modules

[mptntguru]

9. By processing wavs i have meant processing not samples but midi tracks\channels rendered to wavs . But as far as i have understood from your answer I can treats flacs of tracks\channels as samples and I have to export to lossless each "midi channel\track" separately and then create new instrument fxs which will have those lossless files as instruments right?

One guy who consideres himself a guru has told me that it's wrong to render to a song from midi tracks and that first i should get wavs\flacs, correct phase issues and then mix to a song.

[Saga Musix]

Well, you don't need to "treat" FLACs as samples - that is precisely what they are. As such it is completely natural to just load them into a sample slot and use them any way you want in your track. I do not understand what that has to do with MIDI tracks though, maybe because the big picture of what you are actually trying to accomplish is missing.
If you just want to mixdown existing tracks of a song, OpenMPT might be the wrong tool. If on the other hand you just want to use existing track stems and combine them with new work, you are right on track.

One guy who consideres himself a guru has told me that it's wrong to render to a song from midi tracks and that first i should get wavs\flacs, correct phase issues and then mix to a song.

Sorry, that is a lot of mumbo-jumbo that really doesn't have much to do with each other. Maybe that "guru" should explain to you in more detail what he thinks you should be doing according to his experience.

[mptntguru]

There is no secret, and in particular OpenMPT's resampling filters are not of the highest quality - however often objective quality is not what people want.

As far as I suspect rendering in daws is not just simple mathematianl summing with a resampling algorithm. They must do some kinds of micro phase shifting to decrase phase cancellation and phase comb and perhaps it's the reason why daws sound so differently. May be the also apply some hidden dithering.
I know that kontakt does its own resampling but the thing is that libs played from kontakt vst sound differently and worse than if you play libs from a standalone kontakt. The closest in purity to standalone kontakt is digital performer. And openmpt is most close. Sequoia and reaper give noticeable blur. And you can't but hear that openmpt give fuller bass and more details highs well noticeable on acoustic guitars. It's not only my opinion but of a series of people. It should be not just resampling quality but also some other algorithms which are probably absent in openmpt which explains why libs sound like raw high quality wav samples. Of course i turn off all processing in kontakt including modulation.

[manx]

As far as I suspect rendering in daws is not just simple mathematianl summing with a resampling algorithm. They must do some kinds of micro phase shifting to decrase phase cancellation and phase comb and perhaps it's the reason why daws sound so differently.

No. No DAW or tracker or any other serious audio production software will do anything like that by default.

May be the also apply some hidden dithering.

You are not able to hear any difference in dithering at all when using 24bit output. This is thus irrelevant.

I know that kontakt does its own resampling but the thing is that libs played from kontakt vst sound differently and worse than if you play libs from a standalone kontakt. The closest in purity to standalone kontakt is digital performer. And openmpt is most close. Sequoia and reaper give noticeable blur. And you can't but hear that openmpt give fuller bass and more details highs well noticeable on acoustic guitars. It's not only my opinion but of a series of people. It should be not just resampling quality but also some other algorithms which are probably absent in openmpt which explains why libs sound like raw high quality wav samples. Of course i turn off all processing in kontakt including modulation.

I am 100% certain that sound output of VST plugins is not in any meaningful way different, let alone better, in OpenMPT than in any other VST host. If you can actually hear differences (did you do a double blind test for that?), I highly suspect that it might be due to a different setting in samplerate (which the host (DAW) dictates the VST plugin to run at) between the different hosts. Double check that all your DAWs and Kontakt stand-alone run at the same samplerate and also run the VST plugin at the same rate (if per-plugin samplerate is even supported by the given VST host) when comparing.

[mptntguru]

Unlike in DAWs, there is no strict assignment between pattern channels and audio sources. You can place an instrument in any channel you like, so the second column of the entry - the instrument column - is important, because it tells you which instrument plays.

The answer is a bit vague for me. Do you mean that in order to hear for example drums (fx2) instead of piano (fx1) while playing the midi keyboard i have each time to switch to an instument tab and select number 2 at the top, and to hear piano to go back to instrument tab again and select digit 1? Or do such switching in the "plugin" section of patterns tab at the top which seems easier. Still i don't understand why you can't implement automatic choosing the right instrument assigned to the given channel when i select a cell in that channel. It would save efforts definitely.

[mptntguru]

Well, may be its resampling and not phase corrections and still their resamples are closed source and the hue of a daw really exists because i have talked to several sound engineers and they confirmed it exists and audible. Also daws gives out different volume even with all faders on 100% and especially noticeable depending on the given os: alsa vs windows driver for example.

10. It would be nice to have a possibility to set a midi channel number to openmpt channel, thus it would be possible to use one copy of kontakt per 16 openmpt channels which would send midi messages via 16 midi channels. For example I select a sell on channel 2 and it sends via midi channel 2. It's because my midi keyboard sends only via midi ch 1. I could put fx1 with kontakt to 16 openmpt channels and get 16 instruments.

[manx]

Well, may be its resampling and not phase corrections and still their resamples are closed source and the hue of a daw really exists because i have talked to several sound engineers and they confirmed it exists and audible.

Yes, differences between different resampling algorithms can be audible. Which is precisely why I have asked you to run all DAWs as well as the VST at the same samplerate and then compare. In that case, no DAW-specific resampler will in involved at all.

Also daws gives out different volume even with all faders on 100% and especially noticeable depending on the given os: alsa vs windows driver for example.

Compensating for different volume is trivial, and must of course be done before comparing. Even slight differences in volume are easily audible and the human brain tends to prefer louder sounds as "better-sounding".

[mptntguru]

+ also plugin delay compensation feature which should shift phase...