Pending Feature Requests

From Audacity Wiki
Revision as of 10:56, 13 August 2009 by PeterSampson (talk | contribs) (→‎Not yet reviewed: FR forum=>Wiki: advaNoise Removal UI)
Jump to: navigation, search
This locked page is for the use of sysops to transfer feature requests from the Audacity Forum and other sources. They will be reviewed and where appropriate transferred to the Wiki Feature Requests page.
For this page, it is best not to summarise, but paste in verbatim, so the context of the suggestion can be understood.



Not yet reviewed

  • Default recording scale: When I start recording a new waveform, I'd prefer for it to default to the dB scale instead of 0 to 1.0 display. Also, I notice that in the dB scale, the increments are linear, the spacing between 0, -10, -20, -30 and -40 are evenly spaced. Any other dB or VU meter I've ever seen have a more logarithmic spacing. (Plus 3 votes)
    • Koz comments: I would like to set the defaults, too. I'll see if maybe one of the advanced Alpha pre-1.4 versions has that. Nobody is going to change version 1.2. The spacing is linear, but the numbers are in dB which is logarithmic. -20 dB is ten times less. -40dB is one hundred times less. -60dB is one thousand times less. You're right. The zero to one scale will only take you down to about ten to one (0.1) before you can't read it any more. That's only 20dB.
    • Koz notes: "default to the dB scale instead of 0 to 1.0" - I can set that in the pre-release 1.3.7, so it's possible it's already available in the earlier versions like 1.3.5 or 1.3.6 which you can download now. You can install 1.2 and 1.3 at the same time, but you can't use both at once.
    • WC comments: And I add my vote for this too - the dB scale is a much better option for default IMHO.
    • Original poster adds: In Cool Edit 2000 (and I assume CE Pro and Audition), the waveforms displayed in dB run from -oo (how do you type the infinity symbol?) to 0. A -30dB signal barely shows, but so does a -40. In Audacity, if you set the scale to -36, a -30dB signal looks fairly significant while even -36 looks like dead silence. At -96 and higher scales, the same signal looks like it would be quite loud, while it's actually very quiet (I don't run my speakers at max volume). What I'm saying is that I'd like the option at least for the scale to have a log taper like what I'm used to in CE2k and most other audio devices' meters. Whether tied to waveform display or separate, the same goes for the input/output meters at the top.
    • SteveTF comments: I like my sine waves to look like sine waves and I like my measurements to be in dB (though I'm getting more used to -1.0 to +1.0, and that scale is very useful when developing plug-ins). Add my vote.
  • Under Edit > Select, I would like to see more choices than just 'All' or everything fore or aft of the cursor. It would be neat if in that same area a list of all my markers came up so I could also choose something like 'Cursor to Markername'. And/or select Cursor to ... then flyouts to Label Track(s) > Markername(s). This would really help when recording albums and you now want to split the songs quickly. Other selection features maybe cursor to allow more than one cursor? Or something like 'splices'. Splices could be set in just like markers? Then a script could run for batch exports. I'm just thinking I could set all these splice markers, then click Export > Splices, then either one dialogue for batch naming the files (I would edit the tags later individually later) or the dialogue comes up with each splice and I edit all names and tags as processes.
  • PulseAudio: Is there any plans for native supporting of PulseAudio in Audacity? I can't work without PulseAudio, because I use remote (via LAN) small computer with good audio card, due to noise in main PC. Now i run 'padsp audacity' but get strange results in 'Audio device info', etc., and not sure about true 24bit 96kHz really recorded. My OS on both computers is Ubuntu/Linux. My goal is to have sound card on remote computer, so it is works as external DAC/ADC and I get some benefits from this, like low noise and short analog audio cables. PulseAudio is latest and user-friendly system for audio transfer via LAN, so it built-in in Ubuntu and work out-of-box. Fortunately it all is already works, no fail, no lags etc., all fine, except one thing. My experiments shows only 16-bit recording due to not glueless connect from Audacity to driver, as I described here http://www.pulseaudio.org/ticket/443. So I search for native support for PulseAudio in Audacity, PA devices need to be shown in list of available choices, like OSS and ALSA sources (in Ctrl+P -> Audio I/O).
    • SteveTF comments: You could perhaps create virtual device in Alsa to allow Audacity to work with Pulseaudio. This is not something that I know much about, but there is some information on the subject here: http://forums.debian.net/viewtopic.php? ... pulseaudio - If you get it working, it would be interesting to hear how you did it
  • I miss the button in "Normalize" that says [x] Apply the Same Correction to Both Tracks. And yes, I know what the "Amplify" work-around is. I don't like it. (Plus 1 vote)
    • SteveTF comments: I could make a plug-in for you that does that except that I don't know how to make the DC offset correction part, so it would just be "Normalise channels independently / Normalise channels to same value". The first option would be identical to the normalize function in the "Normalize" effect, and the second option would be identical to the "Amplify" effect. How would you want Normalize to behave if multiple tracks are selected? Would you want each track to be treated independently, or all the same? For example, if you select "Normalize to -1 dB", and that requires that the loudest channel of all of the selected tracks needs to be amplified by 2dB, should 2dB amplification be applied to all channels of all tracks? (this is what the "Amplify" effect does). Alternatively, should 2dB amplification be applied to all channels of that track, but different amounts applied (as required) to other tracks?
    • oz responds: "I don't know how to make the DC offset correction part" Yes, well, that's another discussion isn't it? Why do you need that? DC isn't audio--ever--and has no business in an audio channel. That's a work-around and should probably have its own tool. That will make it easier delete when it isn't needed any more. But back to business. The design center for "Normalize" is simplicity. All selected audio tracks are changed so the hottest peak arrives at one selected value either A) using the hottest one, or B) independently. Default is A. If you select B, you are given a warning about damaging the musical stereo image. That's it. One sentence. Any decisions or requirements above that are the providence of Amplify or other effects tools. This business of trying to explain that to Normalize what a lot of people consider "correctly" requires the Amplify tool is a work-around.
    • WC votes and adds: And I agree that the DC offset removal should be an entirely separate tool.
  • Audio Vectorscope: We keep dancing around this, and yes, there are really painful ways to derive this information, but I would kill for an audio vectorscope. Either version; left is vertical and right is horizontal, or the (I think) CCIR/EBU version where In-Phase is vertical and Out-Of-Phase is horizontal. Failing that, a Phase Meter. Bouncing lights to the right are in phase and bouncing lights to the left are out. It's something you can turn on if you need it, like the monitor mode for the record meters. 1.4 will tell you somehow graphically when you're in monitor mode, right? 1.3.5 and earlier doesn't tell you a thing...I didn't check the other two higher versions.
    • Koz adds to his original post: Typically, it would be... View > Show Vectorscope. I was going to put it under Analyze, but a simple version of this can be left running all the time, not launched once for a simple test and then put away.


  • Default Sample Rarte: At present, if Audacity opens or imports, into a new project, an audio file that is at a different sample rate to the default (as set in Preferences), then the project rate will change to that of the imported audio. However, this is only true for the first file that is opened or imported. All subsequent files will leave the project rate unchanged. I find this behaviour to be both confusing and inconvenient. If I have the default sample rate set to 44100, that is because I want my project to use 44100. On more than one occasion I have opened a bunch of files, and only after noticing that the sound quality of the whole project was rubbish, realised that the project rate was set to an unacceptably low rate simply because the first audio clip that I imported was at a low rate. This behaviour can also cause problems for users that have sound cards that only support, or only play back at the correct speed, projects at specific sample rate settings. I would like Audacity to use the default sample rate for projects unless I explicitly change the project rate to another value (for example by using the "project rate" setting in the lower left corner of the main Audacity window).stevethefiddle
    • Gale Andrews responds: The switching of project rate to that of the first imported file (if different from the current rate) wasn't actually working on Mac/Linux if the rate of the file was unsupported, and one of the "fixes" in 1.3.7 is to make that switch always happen. The rationale is that there would have to be resampling if the project rate differed from the rate of the file; but that subsequent files should not change the rate again so that there would be a mix of rates. As you know, all resampling is lossy, and resampling has to be done somewhere if files at different rates are involved, in spite of the general recommendation to keep the project rate at a rate supported by the device. It was the best compromise we could come up with. Audacity (Beta) should if necessary always resample playback from the project rate to the next highest supported rate, so that the sound card is sent a *supported* rate. You can see the rate it's actually playing (or recording) at in the far right of the Status Bar. The Vista Default Format Issue using MME is the exception that proves the rule, where the system is resampling (badly), instead of Audacity doing it. I can't help thinking If the (Windows) world had better device drivers we wouldn't get half the reported problems with speed difference when recording and playing at the same time, and devices that cannot cope with resampling. I agree there is a weak case for not changing the project rate upon file import given how a minority of devices behave in the real world, and your point about project playback quality being destroyed by import of a low rate file is valid. I think a much stronger use case for not changing the rate is presented by people who want a fixed rate for export to CD or DVD. Anyway, the plan is to leave default rate behaviour when importing files as it is, but add a preference (probably after 1.4) to let the project rate remain fixed until the user changes it.
    • STF replies: That would solve the issue nicely, but I think that the default should be to not change the project rate.

Looking at the pros and cons:

Reasons why the project rate should change if the first (and only the first) file loaded is at a different sample rate:

  1. There would have to be resampling if the project rate differed from the rate of the file.
  2. .... I can't think of another.

Reasons why the project rate should remain at the default project rate setting, unless changed by the user:

  1. Changing the rate on account of the first, and only the first file opened is inconsistent with all other situations.
  2. Audacity is designed for high quality audio production and is, in the vast majority of situations, set to a default of 44100Hz (CD standard) or 48000Hz (DVD standard). Up-sampling a file from a low sample rate to either of these rates incurs negligible losses, but down-sampling a project to a low sample rate produces significant losses. Failure to notice that the import of one low quality audio clip has changed the sample rate for the entire project will result in the Exported audio all being resampled to the low rate, causing significant damage to the sound quality, whereas up-sampling a low bit rate sample into a higher bit rate project will result in a larger file size and insignificant loss in sound quality.
  3. A file that has been imported, then exported at a higher bit rate may later be down-sampled to a lower bit rate. Up-sampling a low quality audio clip, and then down-sampling it again will produce a very small degrading of the sound quality. The reverse of this (down-sampling a file then up-sampling it) will produce a large and irreversible degrading of the sound quality.
  4. Many users use Audacity for CD or DVD production, in which case it is an unnecessary inconvenience for Audacity to change the project rate from the default values, forcing the user to manually change it back again. This is also a behaviour that users need to be aware of (and they only become aware of it when they get caught out and ruin a project).
  5. Some sound cards do not handle re-sampling properly, and in these cases the project rate needs to be set and stay set. This may be the "fault" of the manufacturer, but this situation exists for a significant proportion of users.
  6. Vista (sometimes?) has problems resampling audio. Does this issue disappear if the project rate is set to a "compatible" setting?
  7. Since the sample rate switching was not previously working on Mac/Linux, I guess that it would be no great challenge to make it "not work" by design.
  8. The argument regarding "avoiding resampling" does not hold very well, since most/all of the processing in Audacity is done in 32bit, thus necessitating resampling (and dither) for any audio that is at a lower bit depth.
  9. In an open project that has, for example, one track with a sample rate of 44.1 and the project rate is 44.1, if a second track with a sample rate of 16kHz is imported, then the original track is deleted, the project rate will remain at 44.1 (and so when Exported will be at 44.1). However, if the original track is deleted and then the second track imported, the project rate will change to 16kHz. It does not seem logical that in this case the outcome should vary according to the sequential order, since in both cases we are just replacing one track with another.
  10. The bit depth of imported audio is always changed to the default project setting, whether it is the first file to be opened or not.

I agree that some (all) Save Sound File As, of the arguments lack much weight, but on balance I would find the arguments for not changing the project rate to be more persuasive.

  • Ability to use Audacity as a simple WAV editor, or other imported format: Given that 4 out of the 5 primary features of audacity[1] involve editing a single file, would it not be worth making that process easier? e.g. if you load a file, modify something, and press control-S, it should save the file, instead of prompting you to create a project?


  • Default Preferences I deploy Audacity (1.2.6 currently) in a lab environment. Is there a way for me to edit the default preferences for Audacity? The "audacity preferences" text file gets created in ~/Library/Preferences (user space) when someone runs it for the first time. I would like to be able to put an "audacity preferences" file in /Library/Preferences, so that it could be used as the default for ANY/all users, but apparently Audacity does not look in that location like a normal Mac OS X application is supposed to do (placing a prefs file there seems to be ignored by Audacity). For example, I would like to be able to specify the location of the LAME dynlib, for example, in the preference file ahead of time, so that the users are not prompted to tell Audacity where the library file is the first time they run it (a user confusion issue.) [MP3]MP3LibPath=/usr/local/lib/audacity/libmp3lame.dylib
    • Kozikowski comments: It's actually worse than that. The early Audacity used one place and file, the medium Audacities used the one you referenced, and the New Portable Audacity uses yet a third. The developer helpers have the opposite problem. We have to clear the decks for each upgrade and change so that there is no trace of the old one around. Sometimes that's a chore.
  • Auto Complete Equalizer Graph: I created a simple equalizer graph to get somebody out of trouble recently. I think it was a lot more bother than it needed to be. Here's an illustration of an Adobe Photoshop tool. See: http://audacityteam.org/wiki/index.php?title=Image:Curves2.jpg Note that there's only one new data point on the right-hand brightness curve, and yet Photoshop automatically produced a graceful, gentle, useful curve typical of a picture whose natural lighting had actually changed. It did not produce two straight lines and depend on me to painstakingly calculate the new points and put the rest of the curve in by hand. I want the equalizer work window to run like that.
    • SteveTF responds: Since Audacity does not currently support graphical interfaces for effects (which rules out the "CoolEdit Pro FFT filter" type GUI), could this be done using a bunch of sliders?
    • Koz responds: It could, but to figure out where the sliders go, I'd be curled up on the living room floor with my Toshiba adding machine, my Terman's book of six place log tables, number two lead pencil, and nice legal pad. If only we could make a machine to do these laborious calculations for us....
  • More transparent "Sample format" display to left of track: This might be implicit in some questions I've asked before, but I thought I'd say it explicitly: I think it would be helpful if the "Sample format" displayed at the left of each track reflected the actual current bit depth of the audio in that track. What it displays now seems to have some obscure relation to either one's default settings or to the type of audio imported.
    • SteveTF comments: In most situations the "Sample format" displayed at the left of each track does reflect the actual sample format. However, you have noticed that when opening an Audacity project, the bit depth of existing tracks is (incorrectly) displayed as the default bit depth and not the actual bit depth of the track. You suggested previously that you thought this was a bug, and Gale agreed with you. For what it's worth, I also agree. At present, the developers are looking into another issue concerning bit depth and I would hope that when they have fixed that, this issue will also be addressed.
    • Allen McBride responds: You're right... I only decided to make a formal suggestion when I discovered that the actual behavior was more complex still when it comes to imports (the other thread I linked to). I know it's not the biggest deal.
  • Frequency values without selection: I am now developing a plug-in for Audacity. My problem now is on how to automatically get the values on the input or recorded medium without highlighting a portion of it. I mean...is it possible to get the frequency values even if the user will not be able to highlight a portion of the wave?


  • Equalizer Accuracy/Readouts: Where is the readout in the graphic equalizer tool to tell me exactly where I am in both dB and Frequency? If I launch the Spectrum Analyzer tool, I can find an offending frequency to surgical accuracy by moving my cursor over the culprit. 3129 Hz. Then when I launch the graphic equalizer to rid myself of the offending frequency, the best I can do is guess at a point somewhere between 2000 and 5000. I used to be able to get better with China marker/grease pencil and a ruler, but that only works with glass monitors. The China marker will not come off flat panels. I'm using that as an example. I know there are notch tools and I can write my own equalizer, etc.
    • SteveTF comments: The Equalizer does have a grid now (Audacity 1.3.8 alpha), which is a big improvement.
    • Koz responds: Much better than a plain white frame, yes.
  • Long duration recording and auto saving: I need to record audio for days for a project I am working on. I would like to be able to save the audio automatically every hour to WAV file. Is it possible to make Audacity do this???
    • Koz comments: Audacity is a pretty simple program so Exporting and Capturing at the same time can't happen. It's already writing to the hard drive during Capture, so it would have to suspend that process in order to write the Project.
    • Plus one vote from Jaws78: I would like to second this. I want to record long sessions 6hours + and would like to break them up automatically in to smaller segments, so I can just press record and forget about it until i am done.
    • SteveTF comments: I doubt that Audacity would be able to continue recording smoothly while simultaneously writing a 600MB file to disk. Would it be acceptable for the recording to be automatically paused while the file was being saved?
  1. Alternatively:

How about if Audacity just started a new project file each hour, so that when you have "Split long recording projects" selected, it will create a new .aup file each hour? (myshow001.aup, myshow002.aup. myshow003.aup ...) I don't know if this is technically possible either, but it seems more likely than trying to dump 600MB of data to disk at the same time as recording.

  1. Another alternative:

After completing a long recording, how about an option to "Split project to multiple projects"? So you start with a 3 hour recording and instead of: "Save As..." > myproject.aup you select: "Split Save As..." > "Options = split after 60 minutes" > myproject001.aup, myproject002.aup, myproject003.aup Where "myproject001.aup" contains the first hour, "myproject002.aup" contains the second hour, and so on.

  • Need longer filenames: I would like for Audacity to be able to use long (up to 255 characters) filenames

without adding in its own arbitrary codes. I also checked it on a filename that doesn't contain parenthesis, and the same problem still occurs. I start up Audacity and it has a blank project window. Then I import my first audio. file. The name of the input file is "RAH 08 Out On the Edge (itunes)". However, once it's done importing, both the project window and the tracks within it are named something like "RAH 08 Out On the Ed#4330D7". Currently using Audacity 1.2.5 on Mac OS X (10.4), with Intel processor. Update: In a previous session of Audacity, I had saved the project. Then after quitting Audacity, I had renamed both the .aup file and the _data directory to be my "correct" filenames (and matching each other). Today I tried to reopen that project, and it was unable to. Then I renamed the _data directory back to its old name, and the project opened OK.

  • Rename-able and movable projects:It should be possible for Audacity to find the _data directory based on the current name of the .aup file. It should be possible for people to rename both of these and have the project still usable.
  • Several ideas for navigation features: Currently using Audacity 1.2.5 on Mac OS X. I've only been using it a short while, so my experience with it is pretty limited so far. The other audio editor that I've been using is Sound Studio, and some of the navigation features in that program would be nice to also have in Audacity. The following would be nice additions to be able to move the cursor, scroll, and zoom:
  1. PageUp and PageDown keys: these should page the window horizontally left or right.
  2. It would be good to make use of the scroll ball on the Mighty Mouse (or similar devices). At least, it should be able to move the scrolling thumbs of the active window. In Sound Studio, in addition to the left/right motion of the scroll ball doing left/right scrolling of the window, any up/down motion of the scroll ball is used to adjust the horizontal zoom.
  3. Need better control over the vertical zooming of each track. Need it to be easy to keep the zero line centered within the track. Need it to be easy to set several tracks to the exact same zoom amount.
  4. For listening to a single track(-pair) at a time, there should be a way to select which track(-pair) to listen to, with a single button press. Currently, doing this by either the "mute" or the "solo" buttons requires two button presses (turn off that button for one track, and turn it on for the other). Two button presses is too much to require for such a common operation. Perhaps the "solo" buttons could be made smarter so that pressing the "solo" button on one track automatically releases the "solo" setting that might already be on for any of the other tracks.
  • Media INFO I imported a clip into Audacity. What was it? What was the format? Filesize? Where is File > Media INFO? (plus 1 vote)
  • Tone Generator improvements I think a very useful improvement would be to give it a more musician friendly approach. For example rather than generating the pitch of the tone by frequency, why no give the option to choose by musical note too? Add the Music scales to it. Rather than choosing 4000Hz (random number, just an example) you could chose F# or something. Plus with that combined with the tempo you could also give the duration setting to half note, quarter not, whole note, 6th not etc. Sort of like a digital piano. Synth is very popular to use in music so this could make the tone generator much easier to control and use to its full potential in the name of music.
  • Frequency band splitter: I find this feature in Fscape (an xplatform app similar to Soundhack) http://www.sciss.de/fscape/ - to be very useful and would like to see it or a Nyquist script written to do something similar: a panel would appear when Band Splitting function is selected and would allow the user to
  1. define bands as freq/bandwidth -or- crossover freqs
  2. add new bands/crossovers as needed
  3. processes the bands or crossover parameters on a soundfile or a selection of a soundfile
  4. output these as separate files into a selected directory -- similar to splitting soundfiles according to label regions
    • STEVETF comments: Unfortunately this is not possible with Nyquist in Audacity, at least not a one-click automated method, because Nyquist in Audacity can only access one track at a time. It is however possible to make multiple copies of a track (Ctrl+D) and then apply band-pass filters to each track.
    • Anechoic responds: you only need to access one track. How it could work: create a multi bandpass run the track thru it and split out the results (ind bands) into n-number of separate tracks.
    • SteveTF adds: That's the bit Nyquist in Audacity can not do - it can only access (reading or writing) one track. If you process multiple tracks (select several tracks and apply a Nyquist effect), each track is processed in Nyquist independently of other tracks. The audio data is passed to Nyquist in a single variable ("s") and the result that is returned by Nyquist is written back to that track. Audacity currently supports single channel (mono) and 2 channel (stereo) tracks. In the case of multi-channel sounds, the "s" parameter is a vector with two elements (aref s 0) and (aref s 1) which contain the data from left and right channels respectively. You could write a Nyquist script that would split a track into 2 frequency bands by processing a stereo track and returning one band to the left channel and the other band to the right channel. If Audacity supported tracks with more than 2 channels then you would be able to write each frequency band to an available channel, but currently Audacity only supports mono and stereo tracks. Creating multiple duplicates of a track and then extracting a frequency band from each track is a bit tedious to do, but it does work.
    • Anechoic responds: sorry I meant that it created 'n' number of FILES not tracks so it would behave much like how split function works
  • Open-Append: Cool Edit has Open - Append: - it will join a list of audio files in order into a single waveform. I don't think Audacity does this (Plus 2 votes)
    • SteveTF comments: No it doesn't but I agree that it would be a nice feature. In Audacity it would probably be called "Import append".
  • Track hover help hovering the mouse over a track should give the filename (going to Track | Name... is less convenient) (Plus 1 Vote)
  • It would be nice if scaling multiple tracks vertically could be achieved without the need for additional controls. Perhaps one way this could be done would be if manual vertical size adjustment affected all selected tracks. This way, if you want to zoom vertically on tracks 1,2 and 4, you just need to select those tracks and as you adjust the vertical height of one, the vertical height of the other selected tracks would follow suit. What do you think - would that do it? (perhaps with an option in Preferences to switch this behaviour on or off).
  • Embedding controls in a (any) presentation document: I made a PDF of a song book with Acrobat and added a play button to one of the songs which played a recording of that song on my default player (Media Player). I thought how cool would it be if I could add bars that ran the length of the piece of music to pause, stop and play. Then I thought about how if Audacity could be embedded, the tempo of the song could be changed which would really be helpful in trying learn music. Is there any chance of Audacity teaming up with Open Office to provide something like this? Perhaps with Open Office PDF Editor which I hear is in the works?
  • Online (web-based) Audacity: Is there any way that someone can make a version of Audacity that can be used on my website (or here). You see last year my phyics teacher had my class do podcasts (fun right). Of course we had to edit it so we all got audacity. Most of the class had no idea how to install a program onto a computer much less how to use it. I think I can teach them how to use it because I am doing a few demonstrations on how to use forums, wikis, etc. and one more couldn't hurt. I think the problem was that people did not want to install a program that they would use once (or twice) at most. So I think having the program on a website would be more efficient. I think it could work with licencing because you could make people accept a terms of service before loading the program. If someone could make this, give me a link if its already made, how to do it myself, or tell me why this can't be done that would be awesome (well not all the options).
    • SteveTF comments: Audacity is a Desktop application and will not run on a web server. However, you can run it fromm a USB stick
  • Noise Renoval UI" when I want to do noise removal, the window asks me to select the section of noise. when I click on get noise profile, the window closes and then I have to open the noise removal window again. Is that the way it is supposed to work?
    • Koz comments: Yes, but that doesn't mean it's desirable.
    • SteveTF comments: Adobe Audition has a nice feature in their Noise reduction effect - after grabbing the noise profile the window stays open and you can click on a button to "Select all of track". This allows both stages of Noise Reduction to be performed without having to close/reopen the effect. Audacity has the ability to "Repeat Last Effect", but I would like the facility to "Recall Last Effect". (Perhaps Ctrl+SHIFT+R). When using Noise reduction, one could then recall the Noise reduction effect in order to apply it without having to go through the full list of effects to find it. "Recall Last Effect" would also be useful in many other situations - for example, you apply Equalization, but on listening back you decide that it is not quite right, so you Ctrl+Z to undo then "Recall Last Effect" and make a slight change to the setting and apply again. I think this would be really useful for many effects where it is not always easy to get the optimum settings just from the "Preview" (and "Preview" is not available for Nyquist effects).


 

Reviewed but not added - unclear

 

Reviewed but not added - discussing internally

  • Improving the Noise Removal effect in Audacity by adding the "threshold" control that was used in Audacity 1.2.x and offering both a full version (with the additional slider) and a simplified interface that used a single slider for more/less noise reduction. I think that in the simplified interface, could combine both the "threshold", and the "amount by which the noise should be reduced" in a single slider, and fixed values for "attack/decay" and "smoothing" (probably fixed at the current default values that are used in Audacity 1.3.x). At low amounts of noise removal, the effect would be more like the 1.3.x effect, then as the slider was increased it would become more like the 1.2.x effect. Plus 1 vote
    • SteveTF adds: The upshot of all this, is that it would be good to have the threshold slider back, but in addition to the refinements that we currently have in 1.3.x - If the developers think that this makes the effect too complicated, perhaps with the categorisation of the effects menu, there could be two versions, a simple, and an advanced interface.
    • See this Forum topic.


Reviewed but not added - intending to delete

These pending FRs were posted here but on review, appear to be inappropriate for the reasons stated (for example, the Beta already supports this feature). Unless reasons for adding them/more explanations of usefulness/purpose are given, they will be deleted.
Retrieved from "https://wiki.audacityteam.org/w/index.php?title=Pending_Feature_Requests&oldid=12964"