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

• Vn 1.3.3alpha (MAC) when I closed an unsaved waveform, on the pop-up dialog (which asks if I want to save before close) "No" was bordered and "Yes" was preselected. In 1.3.5a there's "Yes" preselected AND bordered. Mostly I'm exporting the waveform and want to close it unsaved afterwards, so I'd like to select "No", in 1.3.3 I just hit spacebar (that's cool!), in 1.3.5 I have to press tabulator twice and then spacebar....maybe it's just an alpha bug, but it's more work. In OS X, you have 2 defaults normally. One is highlighted, the other one is bordered. So, if you press enter (as you do normally if you're used to work in Windows), you're saving before you close (choosing the right one). If you hit spacebar, you don't save before closing (choosing the left one). Every single program I can think of in OS X has 2 defaults (Audition too until version 1.3.5 beta) and it's a feature of OS X I love and miss on Windows (I'm still working in Windows sometimes)...so why not support it?

• Ability to select and delete parts in the waveforms during playback. Playback should just go on, or (if playback is running through the part that should be deleted), start at the cursor that remains after deleting. Now I have to select during playback, stop playback, delete and start playback again. (plus 1 vote)
  • STF writes: I don't see the point of it - it's not much trouble to hit the spacebar to stop the recording before pressing delete, and it's much easier to precisely select a region while stopped rather than trying to select a moving target. Can anyone explain what the benefits would be?
  • Deltacon responds: my problem is this: I've got some vocal recordings from a speaker with breathing noises between the text. I'm listening to the recordings (and check the contents, pronounciation...) and I'd like to delete these breathing noises during playback. Hitting spacebar, deleting and playing again is possible, but it's a lot more work and for me it IS trouble, if you have 7 minutes of text and every let's say 6 seconds a pause with breathing noise. Makes 20 more hits on the keyboard in a minute, makes 140 more hits in 7 minutes.
  • STF adds: Have you tried using the "Truncate Silence" effect in Audacity 1.3.5? With a bit of careful adjustment, this may provide a quicker method of achieving the desired result.
  • Deltacon responds: I haven't tried the truncate silence function until now. And I can tell you it works like a charm. BUT, I still would find it helpful, if I can delete during playback, just because if you're listening to a file and there're maybe 2 takes and you'd like to delete one of them, it would be helpful. Not necessary, but helpful. In addition this feature is still supported by Apple's Soundtrack Pro (which has in my opinion less functions for my needs)...so be better than Apple.

• A different wave drawing style: As far as i understand, strictly speaking just connecting the dots of the sample values with straight lines is not really representing the resultant waveform. The "real" waveform, as produced by an DAC can look quite different and even exceed the sample values. To have a display mode, that could more accurately display the resultant waveform would be good, because it would clear up some misconceptions about digital vs. analog and it would make the visual comparison of waves easier and more reliable. A subsample timeshift feature would be nice too.
  • STF writes: The most accurate representation would be to just show the dots (samples) and not join them up at all. The reconstruction of the analogue wave from the samples is handled by the sound card, not Audacity, so the method used to reconstruct the analogue signal may be different from one sound card to another. This is not something that Audacity can predict, but a simple dot-to-dot linking of samples is in most cases a reasonable representation (approximation) of what the sound card is likely to do. An exception to this is with very high frequencies (close to the Nyquist frequency) at which point the sound cards D/A conversion could be quite unpredictable. However, at normal sampling rates, these frequencies are so high that it is unlikely that the loudspeakers that are used for listening will produce the sound at all accurately. Fortunately these frequencies are virtually inaudible.
  • Nat responds: I cannot really agree with this. The sound card is not "free" to do whatever it wants with the samples, as it should attempt a reconstruction of the originally sampled wave. There is a mathematical "optimum" that the soundcard should try to come as close to as possible. The closer it gets the better it will sound.
  • STF responds: Some examples: All sound cards have self noise (how much? what sort of noise?, All sound cards produce distortion (perfect amplifiers have not been invented yet), All sound cards low pass filter the output (at what frequency? what order filter?), Some sound cards work internally at 16 bit, others at 24 bit, some even higher. A sound card running at 24 bit to render 16 bit data may anti alias the output. Sound cards may or may not use oversampling. Some sound cards use noise shaping, others do not. Should D/A converters use high pass filters? DC offset may be valid data in the digital realm, but it is not "sound". What if the sound card works internally at 48 kHz and is called upon to resample 44.1 kHz data? At low frequencies the conversion can be very accurate, but at very high frequencies the errors can become very high. Consider a "perfect" sound card. I generate a triangle wave at 20kHz and record it through the sound card at 44.1 kHz. I then generate a sine wave at 20 kHz and record it at 44.1 kHz. I play the two recordings back through the perfect sound card. How does the sound card know what shape wave to produce? Try this on a piece of graph paper: Set your X axis to correspond to 48kHz sample rate, and plot a sine wave at 20 kHz (well below the Nyquist frequency?) Look at the dots you have plotted and they look nothing like a sine wave. Could any other wave fit those dots?
  • PS/WC: full discussion retained in Forum but moved to Audio Processing: [1]

 

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.