i have just recorded some voiceover work @ 48khz, using ardour’s “native broadcast” format. unfortunately i’m having trouble having it not sound sped up, specifically on windows machines. i’ve tried (in rezound and audacity, along with ardour’s export) converting from (presumably IEEE 24bit float) to normal 16bit PCM. to make it usable on another windows machine using premiere. the only way to get around the problem is change the sample rate from 48khz to 44.1khz (not converting), so it’s lower (and correct) in pitch. does this mean broadcast infact captures less information than a standard PCM format?
ardour1 seems fairly ambiguous in terms of telling you what bitrate it’s recording in…
broadcast wave is a regular WAVE file with an extra chunk containing the information defined by the EBU.
ardour 0.99’s native BWF files use 32 bit floating point data, which many programs still cannot handle even though it has been a legal format for more than 10 years.
the sample rate is determined by JACK, and is displayed clearly and unambiguously in the very top menu/tool bar of ardour, where is says “SR: 48kHz”.
i suggest that you use sndfile-info on the files you are working with to see what sample rate they are based on.
finally, exported files are never in BWF format, just ordinary WAVE, and use a sample bit depth and format chosen by you during the export process (it defaults to 16 bit signed integer).
haha, no it was even more obvious than that. i’d forgotten that i’d locked the internal clock on gnome-volume-control.