Pending Feature Requests

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.

• Simplify the lowpass and highpass filter interface: Suggestion to remove the "Filter Quality" control from the lowpass filter and highpass filter. This control is almost never used, and it only applies to the 12dB per octave filter (not 6dB per octave as described). In the vast majority of cases, adjusting the value from the default 0.7 will produce worse results than the default values, so why not just remove the option - anyone that understands and really wants the feature will probably know enough to re-enable it. Removing this option would also greatly simplify the code, which to my mind would be a good thing and would avoid a potential point of confusion for a lot of users.

• Recreate cut track: On all audio and video program there is a very cool feature that enable you to split a track and enables you to shift it around. But the cool thing is to be able to recreate the lost or cut track by simply draging the edge of the split track. I don't know the technical name of this action but it is highly needed in terms of editing and shortening songs or music recordings.

• Video Display: the ability to preview any sort of video file so you can edit its audio it doesn't really matter what is the file type, any file type. The useres will be able to create it from there video program than load the complete audio to the video editing program for final phase. Please make me forget about wavelab and soundboot.
  • SteveTF comments:Some video editors will allow you to export the audio track as a WAV file, which can then be edited in Audacity. If your video editing program does not have that facility, you can probably use a format converter such as the free program SUPER from erightsoft to extract the audio from the video file.
  • Original poster responds: I know that method and i use it, but this method hold only if the video has a sound track already. When I have a silent video i cant make synched sounds.
  • SteveTF responds: Would either of these ideas (from the feature request list on the Audacity wiki) suit what you are asking for? 1) Import audio from video files/transport streams: such as AVI/MPEG e.g. by demuxing (13 votes) 2) Import/play video for synchronised soundtrack editing (3 votes) No need for Audacity to edit the video. Alternatively, Audacity could send/receive SMPTE or MTC timecode data e.g. to VLC or mplayer.
  • PeterS: No further response from original poster

• Importing / Exporting multiple tracks: There are many situations where it would be useful to be able to import or export more than one track/file at a time. For example; (1) Adjusting the levels of several tracks so that they will sound the same loudness on a CD - (2) Backing up tracks from a multi-track project - (3) Importing tracks from a multi-track recorder - (4) Importing tracks recorded in other multi-track software - (5)Concatenating multiple audio clips into a single file - (6)Safely transporting projects to other computers (that may or may not be using Audacity). (Plus 1 vote)
  • To this end the following features would be useful:

1. Import multiple files with option of (a) importing end-to-end on the same track, or (b) each onto its own track.
2. Exporting multiple tracks with the option of (a) end-to-end joining into one file (with cue sheet), or (b) each to its own file.

• Applying a fade out to silence, or a fade in from silence using the envelope tool is very fiddly, but is a very common process with multi-track editing. Is there some way that this can be made less fiddly? Perhaps an option in the effects menu? I know that you can already apply a fade (or cross fade), but often the envelope tool is more appropriate because it can be adjusted and is not a destructive edit. Also, trying to recreate the equivalent of the fade, or cross fade effect by using the envelope tool requires using multiple envelope points and takes considerably longer than using the destructive effect. If the envelope tool created straight lines in the normal "Waveform" view it would be considerably easier to use. (plus 1 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.

• I'm getting really tired of pointing people to That Little Black Down Arrow To The Left Of The Track. Can that be changed into some menu icon or some graphic that has a simple name? (Plus 1 vote)
  • WC votes and adds: From the fact that we have to steer folks there so often, one can infer that not many folks realize the LBA hides a drop-down command list.
  • SteveTF points out: I know exactly what you mean, but since you raised the issue, I checked the manual.... It has a name; "Track Drop-Down Menu"
  • Koz coments: And you think somebody who is already lost and fumbling is going to think to look under the little Black Arrow for the Track Drop Down Menu? That's the problem. It should say [MENU] or something. I know there's not a lot of real estate there, but that's where you enable elegant programming or design. It's been my experience that some of the program functions feature a hovering cursor drop-down menu and this isn't one of them, so that's strike 2.
  • SteveTF responds: I agree, a balloon tip would be nice, but for the sake of clear explanations on the forum, you could include an image, though for minimal effort you could just write "click on the track name".
  • Koz replies: That's good. That's much better than TLBDATTLOTT.

• 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.

• Recording meter indictor: On occasions when changing the audio inputs and/or recording devices, it can be confusing to decide whether the absence of activity on the recording meter toolbar is due to no signal, or just that the meter has not been activated. A simple "LED" indicator on the toolbar could "light up" when the meter is active. Also, this could have a dual usefulness as a latching peak indicator (turns red if meter reaches 0dB, or other preset level, and stays red until reset by clicking on it). (Plus 1 vote)

• In the Help menu, shouldn't the "About Audacity" button be at the bottom of the drop-down menu, not at the top?

• 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) of the arguments lack much weight, but on balance I would find the arguments for not changing the project rate to be more persuasive.

 

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.