Old 01-29-2020, 05:53 AM   #1
nofish
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Default v6.03+dev0128 - January 28 2020

v6.03+dev0128 - January 28 2020

+ Envelopes: add actions to move active envelope fader, or selected envelope points, up or down a tiny bit
+ Envelopes: turn off track/send mute button when creating new mute envelope while muted (so it's not double-muted)
+ HiDPI: adjust font antialiasing size threshold for dpi on Windows
+ Render: add support for secondary render format (render simultaneously to wav and mp3, etc)
+ VST3: fix .vstpreset loading (thanks @olilarkin)
# ReaScript: add support for getting/setting secondary render format

Full changelog / Latest pre-releases
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Old 01-29-2020, 06:29 AM   #2
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The ReaScript/API support for setting the render format is actually not working in this build, will be fixed for the next one.
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Old 01-29-2020, 07:05 AM   #3
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MIDI editor ruler font is still not antialiased as it was in R5...
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Old 01-29-2020, 07:34 AM   #4
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Take envelope - dont dispays value while drug up down (screen from stable vers where it shows).
Does its possible to add option - double click on envelope opens mini window to set value? Its possible be much better to fast set value in some situations.
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Old 01-29-2020, 09:22 AM   #5
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The text in the "Take Media Source" field is still not visible when using MacOS dark mode:

https://stash.reaper.fm/38180/Screen...20.14%20AM.png

An item is selected and there is technically text there, but you have to highlight the text to see it.

Other areas of the Item Properties Tab seem OK.
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Old 01-29-2020, 09:34 AM   #6
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Default Timecode Bug

I found a timecode bug in the latest release and pre-release.

The timecode at which any WAV file was recorded will be lost if the user renders the audio file to different sample rate than the original file.

Steps to excute bug:
1) Create a session with the sample rate of 48k

2) Place a WAV audio file at any timecode and glue the audio item so it write the HMSF timecode position into the audio file.

3) Open Item properties and make note of the HMSF position.

4) Render that WAV audio file to a different sample rate than the current session. For example 44.1k

5) Drag the rendered audio file back into your session and run the following action: "Move to source preferred position (used by BWF)"

The expected outcome would be that the audio file moves to the same HMSF timecode position that it was originally printed but if we look in item properties the HMFS position is incorrect.


It is extremely important that timecode position is correct across every sample rate, especially when working in film or television. If the mixing engineer for a film or show decides to work at a different sample rate than the one in the user has printed their file at, then the timecode will be incorrect and the audio will be out of sync with video.

Perhaps the solution would be to put a checkbox in render options to retain HMSF timcode position no matter what sample rate the audio file is rendered to.

Last edited by srdmusic; 01-29-2020 at 05:18 PM.
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Old 01-30-2020, 04:48 AM   #7
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@srdmusic
Which was the last Reaper-version, where it still worked?
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Old 01-30-2020, 05:08 AM   #8
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Quote:
Originally Posted by srdmusic View Post
I found a timecode bug in the latest release and pre-release.
I can't reproduce this. Exactly how are you doing the render in step 4? If I use File > Render and render selected media items, the embedded timecode is correct.
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Old 01-30-2020, 08:22 AM   #9
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Default Video format not remembered on closing and opening render window

Think I've found a bug that was introduced when 2nd render output was added.

The Video format not remembered on rendering and then closing and opening render window.

I set it to say MasterMix/Time selection, QT format and when I come back to the window again, it's been set to web format.


This is on Windows 64 build.

Thanks!
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Old 01-30-2020, 09:29 AM   #10
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Quote:
Originally Posted by schwa View Post
I can't reproduce this. Exactly how are you doing the render in step 4? If I use File > Render and render selected media items, the embedded timecode is correct.
Thank you very much for checking into this Schwa, I am almost always rendering stems (selected tracks) Bounds: Time selection.

Last edited by srdmusic; 01-30-2020 at 09:35 AM.
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Old 01-30-2020, 09:31 AM   #11
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Quote:
Originally Posted by mespotine View Post
@srdmusic
Which was the last Reaper-version, where it still worked?
I'm actually not sure it's ever worked properly on my end. I just went back and tested an older pre-release and it's also producing incorrect record position. A I tried an older portable install is also producing the same result. I tested this on one Intel and one AMD windows machine as well as a mac laptop.

Can you produce the same unexpected results on your machine? If no one can reproduce the bug on their end then it might be my rig or some combination that I'll have to test further.

Last edited by srdmusic; 01-30-2020 at 03:54 PM.
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Old 01-30-2020, 03:54 PM   #12
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Quote:
Originally Posted by schwa View Post
I can't reproduce this. Exactly how are you doing the render in step 4? If I use File > Render and render selected media items, the embedded timecode is correct.
If no one can reproduce the bug on their end then it might be my rig or some combination that I'll have to test further.
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Old 01-30-2020, 10:47 PM   #13
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@sdrmusic

When you import the downsampled file into Reaper, is the project still at 48kHz, or have you gone down to 44.1 in the project too?
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Old 01-31-2020, 05:00 AM   #14
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Quote:
Originally Posted by ReaMike View Post
@sdrmusic

When you import the downsampled file into Reaper, is the project still at 48kHz, or have you gone down to 44.1 in the project too?
Basically the problem is that I like working at 48k or higher but some of my clients like working at 44.1k. I get complaints that when I export a 44.1k file from my 48k session the files do not import into my clients 44.1k session at the intended time-code.

Funniest thing though, I did another test today and I can't get it reproduce the same issue when importing back into Reaper on v6.03+dev0128.

If no one else can reproduce the error, then the bug has either magically disappeared or there is something weird with the way my client is importing my audio into their Cubase session.

I'll try to investigate further.
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Old 01-31-2020, 05:06 AM   #15
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Quote:
Originally Posted by srdmusic View Post
Thank you very much for checking into this Schwa, I am almost always rendering stems (selected tracks) Bounds: Time selection.
I'm not able to reproduce using this render method, either :/
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Old 01-31-2020, 09:23 AM   #16
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When I render to both WAV and FLAC, and then diff the files, they virtually null. However, if I render to MP3 (LAME; CBR 320 kbps), then the difference seems too large. But if I then nudge the MP3 exactly 1 sample to the right, then the WAV and MP3 do null again, so it would seem that MP3 is rendered with a 1 sample offset.

This is likely not new, but I noticed because you can now render to both WAV and MP3 at the same time, so I thought I'd mention it here anyway.
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Old 01-31-2020, 09:54 AM   #17
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Some plugins can rename their automation parameters (like Kontakt). Reaper is smart to refresh this when we reopen the envelope panel. However, if we already had a parameter assigned to show as track envelope, this is not refreshed then.

Track controls are refreshed, but not envelope name on the track (it was not aliased). It requires completely clearing the envelope until it uses the changed automation name. Can this instead be refreshed when automation parameter name changes (unless we aliased the name of the parameter, of course), please?

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Old 02-05-2020, 09:26 AM   #18
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Quote:
Originally Posted by Tale View Post
When I render to both WAV and FLAC, and then diff the files, they virtually null. However, if I render to MP3 (LAME; CBR 320 kbps), then the difference seems too large. But if I then nudge the MP3 exactly 1 sample to the right, then the WAV and MP3 do null again, so it would seem that MP3 is rendered with a 1 sample offset.
FWIW: I take this back, because if I convert the MP3 to WAV (using Winamp), and then compare this converted WAV with the original WAV, then the do vitually null. So I guess the 1 sample is actually subtracted when importing the MP3, which has nothing to do with this dev version.
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