I had to do an interview (for a radio show) with a couple instead of just one person. In order to capture all three voices I used my iPad plus RODE two-lavalier-mic setup (Hindenburg Field Recorder) for the two interviewees, and a handheld Zoom for my own (interviewer) voice. The Zoom was recording direct to mp3 and the iPad to wav.
So I thought I was being clever: I’ll just load both files into Audacity, and the strong signal from the Zoom will overlay the weak background signal from my voice coming over the lavalier mics clipped to my interviewees, so I’ll have decent quality all round. And I can adjust the relative volume levels of interviewer and interviewee voices in mixdown, because they’ll be on separate tracks. What’s not to like?
I haven’t tried this before. It quickly became obvious that it was a Bad Idea, and in fact I’m in a bit of a bind now.
What happened – and I don’t really get it – is that the time base for the two recording devices seems to be ever so slightly different. The mp3 track, over the same thirty minutes of wall clock time (first section of interview) is slightly shorter than the same thirty minutes of the wav track. I aligned the start points of the two tracks carefully, and at first all is well, we’re in synch and it sounds great.
But after only a few minutes, I start to hear a subtle “phaser” effect, and pretty soon we have serious echo (or “pre-cho”) as the mp3 track is quite audibly ahead of the wav. FFWD to 15 or 20 minutes in, and it’s almost unlistenable because of the babble of out-of-synch voice.
So now I have a real mess. The interviewees’ voices are just barely audible in the background of the mp3 (zoom) track. My own voice is audible in the background of the wav (iPad/Hindenburg) track – but not loud enough to use. So this out of phase problem cannot be ignored. I have no idea how I’m going to fix it! This was a great interview and I’m just about in despair over how to rescue it.
I’m only a shallow user of Audacity, no real expertise, I just edit and adjust levels usually, mixing down to mp3. Is there a way to establish a marker A and B on Track 1, then a marker C and D on Track 2, then tell Audacity to stretch Track 1 so that the content between A and B occupies exactly as much playback time as the content between C and D?
Or (hoping against hope) is there any super smart Audacity filter that aligns the amplitude envelope of Track 1 to the envelope of Track 2? The envelope is very recognisable (same features occuring, though with different overall amplitude, on each track). I can visually identify the alignable features, and visually perceive the drift between the tracks.
I had no idea this could even happen. I thought that the timeclock on digital recording devices was trustworthy. I mean, the Zoom marks time down to the hundredths of seconds. It never occurred to me that two devices could record the same audio source and come out with different timebases – off by more than a second if you listen long enough!