Throwing something else out there: looking at d_soundfiler.c, it looks like it wouldn’t be that hard to add a second outlet that could send out the number of channels and samplerate of the loaded file. As far as I can tell, the number of channels is already read, so it involves reading the samplerate. This would allow for the calculation of the correct playback speed for samples irregardless of source & playback samplerates.
WAVE: The samplerate is the 4 bytes after the number of channels in the WAVE fmt chunk. Ref: http://www.lightlink.com/tjweber/StripWav/Canon.html <http://www.lightlink.com/tjweber/StripWav/Canon.html> AIFF: The samplerate is the last field in the COMM (Common) chunk. typedef struct { ID ckID; long ckSize; short numChannels; unsigned long numSampleFrames; short sampleSize; extended sampleRate; } CommonChunk; Ref: http://www-mmsp.ece.mcgill.ca/documents/audioformats/aiff/Docs/AIFF-1.3.pdf <http://www-mmsp.ece.mcgill.ca/documents/audioformats/aiff/Docs/AIFF-1.3.pdf> -------- Dan Wilcox @danomatika <https://twitter.com/danomatika> danomatika.com <http://danomatika.com/> robotcowboy.com <http://robotcowboy.com/>
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