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Okay, I'll start this out!
1) I've been using Audacity 1.2.3 for several months and it's been great. But for some reason, which I don't have a clue about, suddenly upon playback, Audacity makes voices sound like Alvin (of The chipmonks). Even my own voice, recorded via line in/from Voice Editing software, is Alvin! Music recorded via webcast radio is also like The Chipmonks! The only change I see is that "Recording time remaining" went from 460-something down to under 200. I've tried uninstalling (and getting rid of all files or folders dealing with Audacity), then defrag and restart before installing Audacity again. All the same situations, including "Recording time remaining" came right back.
Probably recording two tracks at two different bit rates, or dragging other audio files into a project that were recorded at a different bit rate.
2) Since I'm not a musician, I don't understand what the Speed, Tempo, Pitch . . . settings ought to be, nor what they were as default. Is there a way to return Audacity to *all* of its default settings and, by the way, what ought they to be?
Thanks SO much--Arlene
Audacity SpeechCast Processing
Q: I had just started toying with mobile voice recordings, personal musings, presentations at work, and lectures at a course I'm taking. For the personal notes, I have control over the environment and intelligence-to-noise ratio. However, in the other environments, I am limited in this regard, as I cannot stand next to the presenter/lecturer with my Zen in order to get better quality voice recording. :)
I have today for the first time begun looking for good audio processing tools. After playing with a few I am begining to settle I think on Audacity. However, I have no idea what to do for voice processing. I will outline what I have learned so far, and where I think more info would be great for people in general recording speech sound-lets for re-distribution or private records.
My Process to Date
1) My portable recorder (like most, has a limited mic and gain there, and also quality/sampling-rate i would imagine) generates Wav Files, which I import to Audacity.
1.1) This is a stereo, 16kHz, 32-bit float audio (as from the box info on the left of the audacity window)
2) I then immediately save the import as an Audacity project, copying the source wav into the audacity project. I then remove the source wav file which is now redundant (though I still have a copy of this original info on a backup medium until I am happy with the processing of the data)
3) I select the envelope tool and 'widen' it, in preparation for normalisation (to get better normalisation, or increase in volume).
4) Next I select all the data and perform normalisation (audio level of speaker at distance of few metres is low, so this increases the audio level)
5) Unfortunately, the background noise was close to the level of the speaker most of the time, so I select 'quite' pieces between words/sentences and use them as 'noise training data' with the "Get Noise Profile" option of the noise removal 'effect'.
6) Armed with this, I select data for a range to either side of this training data, and apply the noise filter. I repeat for the entire data.
7) I revisit the file searching for gaps in the speech, noises that are not intelligence, etc, and silence/cut them.
8) I have also applied the 'Click Removal' effect, but I am not sure it has done much.
9) My audio recorder (a zen vision:m portable thingy) has a drive in it which spins up a various intervals. Then there is a 'click' and the spin 'whine' has stopped. I presume this is just writing the buffer to disk, but hearing the disk in the recording is annoying, especially with low volume speakers. I am manually looking for these points (which, like people coughing, is given away by a quick 'spike' in the waveform representation). Is there a better way ?
10) There is some 'hiss' and other high pitch noise in the background, how can I get rid of this ?
What I think is needed
1) Some more information specific to speech processing.
1.2) I have read articles hinting at unrequired frequencies or frequency bands, bass mostly. I need to look deeper into this. What are they ? How can one apply them in Audacity ? Are there dangers in losing intelligence ?
1.3) How to use 'filters' to remove data outside the range of the human voice. Can a specific speaker be profiled via a short speech segment, and use this as a mask to remove everything else ?
1.4) What are good audio file settings for speech ? I am referring specifically to sampling sensitivity and rate, stereo/mono, etc. How can one easily re-sample an audio file with speech only in order to make it a smaller file yet with all the intelligence still intact ?
I am sure I am missing loads more good tricks for processing speech data. So please chip in. This is a cool application, good info on accomplishing various tasks with Audacity would be a great help to me, and lots of others from what I am reading online.
With the advent of the multitude of portable digital audio players and recorders and thus the desire to produce general SpeechCasts, we have entered a new era where the general public, not knowing about the Signal Processing tricks required to sanitise speech well, will want to do just that.
Any tutorials, automation, etc would be very very helpful in my opinion.
I will report back if I find anything else of interest.
Q: I'd like to know which resampling algorithm does audacity use. I`m studying resampling for my thesis and I`m testing Audacity resampler` influence on perceived audio quality..
A: Audacity uses the resampling algorithm from Julius Orion Smith's Resample project. Audacity also contains code to use libsamplerate, but we can't distribute librample with Audacity because of licensing issues.
See these discussions for more information on our choice of resampling algorithms: