I’m running Audacity on OS X Yosemite. Recently, I split a long project into 3-minute tracks using the Regular Interval Labels feature. I then exported the tracks using the WAV 16-bit PCM preset. I was surprised to see that when playing back the WAV files, there’s a tiny but detectable “skip” (it’s not really a gap) between the tracks. I would expect the transition between tracks to be undetectable. Any ideas why this is happening?
I’m aware that when burning a CD, the track times must be a multiple of 1/75 second, but this shouldn’t apply when playing back the WAV files directly on a PC or network music player, right?
Please be aware that Audacity 2.0.6 and earlier do not officially support Yosemite.
2.1.0 will provisionally support Yosemite. If you are not already doing so I recommend you subscribe at the bottom of http://audacityteam.org to receive an e-mail when we release new versions of Audacity. To subscribe, enter your e-mail address and click “Add”. Then you will hear when 2.1.0 is released, which should be soon.
What player is playing the tracks (make and version number)? If that player does not support gapless playback then yes, multiple tracks may not play seamlessly - the player may not buffer the stream properly between tracks as the stream stops and starts.
ITunes “should” support gapless playback, though after Apple removed the “Gapless Playback” check box in iTunes 11 there seem to be many reports that gapless playback does not work as well as it did.
Thanks for your reply, I was not aware that Yosemite is not officially supported yet. That being said, it has been working perfectly for me and I don’t think this gapless playback issue is related to Yosemite.
To answer your question, I’m playing the WAVs on a Cambridge Audio NP30 network player that does support gapless playback (confirmed with other files).
I’m guessing (?) I might have to make the track split points a multiple of some value, but I don’t see why this should be necessary unless I’m burning the tracks to a CD (in which case the 1/75 second rule would apply as per my last post).
Other files in what format, bit depth and sample rate, written by what applications?
Are you loading files on the player or network streaming them to or from it? Networks have lags that will affect buffering.
For a case like this you should seek feedback from Cambridge as well, and give them a couple of example files that don’t play gaplessly on their hardware: https://techsupport.cambridgeaudio.com/hc/en-us.
Does that Cambridge player control app support marking its own play points or bookmarks in a single file? It could be a solution if that is the only player you want to use.
Mainly ALAC (Apple lossless) and FLAC files; I know, not the same thing as WAVs. I’ll have to do some further experimentation to determine whether it’s the player…