Mark Waddingham" wrote:
command playSound pSoundFile
if the environment is "mobile" then
play pSoundFile
else
set the filename of player "myHiddenPlayer" to pSoundFile
play player "myHiddenPlayer"
end if
end playSound
-----
Mark: Welcome back to "After The Conference Land" wish I could have been there.
play fork code: Thanks, perfect… we are already using this model for actual
exposed players w/controls for the user since we have to fork to a mobile
player in iOS/Android and the native LC player on desktop anyway, so this
follows that same paradigm, only even easier. In fact we might adopt this
instead of an exposed player and only give the user the option to stop or
start, even for a long recording. I'm not close to user expectations for
control over audio play. The younger set here run all day with earbuds in… I
don't… Even in iTunes all I ever do is stop or start what I'm listening to. I
don't think there is a strong use case to scrub forward or back, but I could be
wrong.
The dictionary offers these are related props in entry on "play" property:
looping, dontRefresh, playRate, showSelection, frameCount, playLoudness,
callbacks, currentTime, playDestination
Can we extract the currentTime from just a "play sound" in progress or that was
stopped? Is suspect that level of control is only available for the player.
SIZE ISSUES:
I just saved the same 49K mp3 file as WAV with the same sample rate and bit
rate settings as wav. The difference in size was even more dramatic than I
expected. WAV was 678k! We are not talking here about short beeps or quick
"bird tweets" or midi type loops, but relatively long (for apps) 10-30 second
voice instructions.
So if one is to add e.g. 100 , 15 second sound files to your app package, that
would be:
~5MB of MP3's
vs
67.8MB of WAV!
OT -- RECORDING: I'm a newbie on this mobile delivery platform. We do have our
high end Sennhauser → Edirol recording system for the important work, that
goes to web, where I run files carefully through dynamics processor, tube
equalizer, scrub the high range, normalize and save with dither… etc., but I'm
looking here at at Q and D production process that still meets the
requirements. Any advice appreciated.
Input settings in Adobe Audition set to Sample rate 16000 hz, mono, bit depth
to 24. 14 seconds. Made with an inexpensive USB Plantronics USB headset/mic:
same one I use for Skype. Anything lower that this starts to sound terrible.
Only sound processing was to sample the hiss on the floor and remove that with
noise reduction and export with same settings as input.
The wav has a minutely better quality than the mp3… which one would expect. But
for voice I'm not sure the difference will be perceived by the user. You have
to listen to them side-by-side to "get critical"… for those interested, check
out these two sound files… a 14 second "instructions" test.
http://wiki.hindu.org/outgoing/instructions.mp3
http://wiki.hindu.org/outgoing/instructions.wav
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