If not by tempo shift, then you could still edit the sampling rate manually by the time ratio 0.99992806, so 44100 Hz becomes 44096.8276 (or round to the precision allowed) and the duration changes to 115 min, 50.5 sec, for example then resample that non standard rate to 44100 Hz in any tool you choose. Any tuning variation would be negligible and would presumably be preferable to tempo-stretching artifacts. (This based on my recollection of the options in CoolEdit, its predecessor)
This should absolutely be possible, but my experience is with an older version(s) of Audition. Also, I barely used the CS versions, so things might be different with newer versions.Assuming you already found the "Time and Pitch" effect, I think the following settings will suit your needs (with some instructions varying depending on program version):1) use Constant Stretch;2) select "Adjust Tempo (preserve pitch)";3) set Ratio 99.997 -- adjust this number until the desired duration shows in the "Length" box;4) select High Precision Quality;5) Time and Pitch settings -- select Use Appropriate defaults (you may want to manually adjust Splicing Frequency and Overlapping values, which is trial and error. Read the included docs).Good luck!
I have tried using these methods, but sadly, it's limited (the % only allows xx.xxx).And no matter what i do, it sounds a tiny bit, "robotic".I only want to stretch and resample with high precision, i want no pitch change or anything else.
If all else fails there may be another program than Audition. I have heard Soundtouch many times but I have not used it myself. Perhaps someone else at HA knows more on this.
"C:\Program Files\bin\sox.exe" input.wav output.wav speed 0.99992806272930005035608948996473
How good is the resampling done there (is it transparent and high accurate etc)? How big number can they work with (Float, 32bit, etc?)
SoX Resampler is extremely good - one of the best out there. It has been tested extensively for aliasing etc and comes out extremely well. It's also been made available as a Foobar2000 DSP plugin because it's good yet light on resources and can be set to resample specific rates only.SoX uses floats internally.From a search (Sox Video file), I believe it's necessary to demux using ffmpeg or similar (or throw the AVI into Audacity to extract the audio track(s) then Export from there to WAV.You can probably pipe the output of ffmpeg into SoX too.
Okay well, it "worked" with RAW, but i get a high Click in the beginning.But it seems to atleast speed up/down correctly.
Quote from: zerowalker on 07 June, 2013, 05:37:08 PMOkay well, it "worked" with RAW, but i get a high Click in the beginning.But it seems to atleast speed up/down correctly.I've noticed that video files don't always have a precise match between audio and visual streams (a fraction of a second difference - though usually I assume this is down to frame-boundaries in non-PCM audio like AAC or AMR and frame boundaries in visual stream).I might try dragging the video file onto Audacity instead of Audition to see if there's a difference in audio stream extraction.If there's a click at the start, usually there's something like a sudden jump in sample value (not starting at 0). The usual method to tame that is to apply a very short fade in (and poss fade-out). For example select from time zero to about 0:00:00.05 and fade in. Around 50 milliseconds is usually enough and won't be noticeableIt is usually good practice to fade at start and end of samples you might want to run together (alternatively, cut samples at zero-crossing points using the function to adjust selection to zero crossings or cross-fade and overlap when mix-editing to replace a short section of audio).
Did you try "resample (preserves neither)" in stretching mode? I've used Audition to tune samples used in Gigastudio. A plain old resample will sound like turning the pitch control on a turntable with no processing artifacts. A 1/2 second total run time runtime change in 2 hours will be undetectable pitch wise UNLESS you're attempting to mix it with another audio source that WAS synchronous initially. At that, the beat rate will be lower than any musician can tune an instrument. If I did my arithmetic correctly, your A 440 would be 439.968. Having tuned musical instruments, I can tell you it's difficult to achieve that precision reliably.Are you trying to match the audio to video? Otherwise it's hard to imagine why a half second is that important across 2 hours. If there is no lip sync to maintain, you could add or delete a few milliseconds during silences and no one would be the wiser and since it is processed not at all, there will be NO artifacts. None.G²
Off topic to this a bit, but i have tried using Sox for changing PAL to NTSC, slowing down and keeping pitch.And, i don´t know if i do something wrong, but i can detect artifacts pretty easy.I tried using Stretch and Tempo.Tempo did alot better.But i also tried with Adobe Audition, and that was was flawless to my ears.EDIT:Okay Audition was not flawless, it didn't produce the artefact, but instead, it doesn't even sound the same, the pitch is wrong;S
SoX attempts to do a useful job here, so probably okay for speech, and to hear what a song might sound like if played at a different tempo, but, given the the simplicity of the algorithm (WSOLA), it's not expected to be perfect. Izotope has something more sophisticated in this area, called 'Radius' IIRC.
In movies, filmed at 24 fps, PAL (analogue) TV systems have often simply changed the playback rate to 25 fps (50 interlaced fields per second), which introduced a pitch shift to the audio making music shift by a little less that a semitone. I'm not sure if digital video broadcasting systems are frame-rate agnostic yet. I suspect not due to support for legacy CRT televisions with fixed scan speed via set-top boxes.I won't address the Stretch or Tempo on PAL to NTSC conversion. For much music, 2 or 3 semitones of pitch adjustment or the equivalent stretching/tempo change sounds OK, but certain tracks begin to warble badly. Sometime it's acceptable to change the pitch a little less then speed-up or slow-down the track for an extra semitone or two despite an accompanying tempo change. (That's why I suggested a speed change and resampling to ensure ideal quality for your original question). Some of this will depend on the algorithm and characteristics of the music, and some of the proprietary algorithms may be superior to open algorithms. Audacity has a bunch of plugins that may be different to Sox's (which I think uses SoundTouch).It may be that despite the inaccuracy of Audition's stretching you could get as close as possible then make the final precise adjustment using SoX's speed function, thus combining a good quality pitch-preserving tempo-changing algorithm with a precise speed change that will be used for less than 0.1%, thus causing negligible pitch variation.Returning to the original question in this thread:What you posted of the "error message" (more of an informational message) indicates that certain of the new sample values were calculated to be greater than full scale. It sound ds like it might only be a single sample.I hadn't come across this in my test audio (because I had about 6-8 dB of headroom to full scale). The error message suggests that your audio is near full-scale and that the re-positioning of sampling points has produced one or two samples that are beyond full scale (clipping). This is always a potential risk with any filtering or re-timing operation and is akin to the phenomenon of inter-sample overs and is also related to the Gibbs Effect.You have a few choices, all of which are likely to be benign in this case• allow the single sample to clip, which introduces clipping distortion. For this operation it's likely to be a minimal clipping distortion (a tiny change in amplitude), and for one or two consecutive samples, it's usually impossible to ABX as proven by the fact that it frequently occurs in lossy codecs' representations of music CD audio.• apply the suggested attenuation command (i.e. negative gain) to prevent clipping. A change of less that 0.5 dB is probably negligibly different to the human ear.• apply something like a lookahead limiter that will leave all audio unchanged except in the close vicinity of the clipped peak, where a soft limiting curve will be applied to produce minimal harmonic distortion and tame the peak. Foobar2000 has Advanced Limiter for this purpose. Audacity and Audition have their own under different names.
The work i am trying to do, has to do with syncing video to audio.And i am a perfectionist when it comes to this, it Must be correct, both the video part and audio part. It's the hard frustrating way, but it's the True way;P
Quote from: zerowalker on 08 June, 2013, 02:15:06 AMThe work i am trying to do, has to do with syncing video to audio.And i am a perfectionist when it comes to this, it Must be correct, both the video part and audio part. It's the hard frustrating way, but it's the True way;PIf you are a perfectionist how the the audio and video get out of sync? Answer: the audio and video weren't locked during record. The "True way' is RESAMPLE. What you're attempting is incorrect in theory and practice. Think of it this way. Obviously the audio and video WERE in sync when it was laid down so the audio has been 'shrunk' because the playback sample rate (now locked to the video) is faster than the original record sample rate - that's why it's wrong now. You want to restore the time so you have to recalculate what the sample rate WOULD HAVE BEEN if it were locked during record which is exactly what 'resample' does.I work in a Hollywood post house and your problem - for the same reason - pops up now and again and it gets fixed -sometimes scene by scene - the way I'm telling you. It always traces back to somebody who thought they knew more than they actually do.Someone mentioned the speed difference between PAL and NTSC. That peed difference is 25/23.976 - about 4.17% - more than 2/3 of a semitone. Your amount of change is 13900/13899 - less than 1/10 of the difference between NTSC B/W and NTSC color. Before pitch correction was practical the Europeans simply listened to American video 4% fast.G²