[FFmpeg-user] 5% of audio samples missing when capturing audio on a mac

[Carl Zwanzig]

On 9/22/2020 8:29 AM, Edward Park wrote:
> I might be making up the history behind it, but 44.1kHz was basically
> just workable, with 20kHz assumed to be the “bandwidth” limit of sound
> intended for people to hear, 40kHz would be needed to encode sound
> signals that dense, and the extra 4.1kHz would help get rid of artifacts
> due to aliasing - and probably the biggest factor was the CD.

My recollection is that you're substantially correct- tradeoffs of number of 
the bits on a CD, human hearing (most people can't actually hear up to 
20kHz), however....

"The official Philips history says this capacity was specified by Sony 
executive Norio Ohga to be able to contain the entirety of Beethoven's Ninth 
Symphony on one disc.[25] This is a myth according to Kees Immink, as the 
EFM code format had not yet been decided in December 1979, when the decision 
to adopt the 120 mm was made. The adoption of EFM in June 1980 allowed 30 
percent more playing time that would have resulted in 97 minutes for 120  mm 
diameter or 74 minutes for a disc as small as 100  mm. Instead, however, the 
information density was lowered by 30 percent to keep the playing time at 74 
minutes"
(which some of the things I was recalling, too)


As for 30000/1001- that's an artifact of NTSC analog television trying to 
fit color information into a b/w signal and then later applying SMPTE 
timecode to the resulting frame rate. There's a good explanation at 
https://en.wikipedia.org/wiki/SMPTE_timecode#Drop-frame_timecode

z!