On Sat, Mar 6, 2010 at 12:32 PM, jtd <[email protected]> wrote: > Ok. From the response one can conclude two points > 1) Closed formats > 2) The consultant is sticking to a single tool when he has undefined > jobs. > 3) The consultant does not understand the real issues of audio > processing using computers. > 4) This requirement is not for professional high definition audio > recording, - the type you would have in a SciFi movie or a modern > heavy metal band - but the usual government ration bhashan editing. > > 1) Dont use closed formats EVER, irrespective of any arguments. If you > need quality use analog tapes (and incur the cost of maintaining > those - remember it is very expensive). For archiving use raw 196Khz > sampled data. For other archiving purposes use flac. For consumption > uses ogg. > All parameters of an ogg encoding can be customised. So depending on > the audience (mp3 player / FM radio / Internet radio / hifi etc ) > you can set the sampling rate, sample size etc providing suitable > compromise on quality v/s size/bw. > Most important - use of closed formats will make everyone incur a > penalty while playing back. If the government wanted to stream a > closed format, they will directly be paying a fat sum for closed > streaming codecs, instead of using VLC / Apache+ icecast/ shoutcast / > some taken-for-granted FLOSS tool chain. > > 2) When you have undefined batch jobs like converting multiple input > formats / quality to a single predefined format, use sox (and ffmpeg > for video or audio container). > One has to spend a few hours understanding the workings o these tools. > Even if one builds a gui, this requirement would not go away - > undefined problems have a way of styming everything except the CLI. > > 3 and 4) One does not record sitting on your desk. You require an > anechoic (or a room with a well defined acoustic print) and fixed > high performance microphones. And you would have a table load of > equipment to check dynamic range, TIM distrotion, frequency response > etc. > Also clock jitter and drift shows up as distortion. Clock jitter is > never checked in a pc and quite a bit of drift occurs as the ambient > and chip temperature changes. Thus pcs will have markedly different > distortion depending on the difference in the source data clocks and > the sound card clock. The older sound cards used an external crystal > (which was good). The new onboard sound uses the motherboards 14.318 > xtal and a pll to generate the 24.576 Mhz clock. They are hence > exposed to the jitter of he 14.318 Mhz signal (besides their own). > Hence you use a PRO quality sound card like the delta 1010 with "word > clock I/O for sample accurate device synchronization" or an external > DSP box. > > Finally live recording / dubbing of audio at a desk is used mainly for > porn, shaadi-happybirthday scenes. "quickly fixing recording errors > by punching in corrections on-the-fly " a must for my kids first > classic rendering of "Three little kittens". >
In addition see http://lists.linuxaudio.org/pipermail/linux-audio-user/2010-March/067825.html A nice guide for Advanced Audio Equipment for Linux: http://lists.linuxaudio.org/pipermail/linux-audio-user/2010-February/067030.html But, do see Ubuntu eeepc: http://lists.linuxaudio.org/pipermail/linux-audio-user/2010-January/067001.html This discussion has been going on at http://www.ilug-cal.info/pipermail/linux-discussion/2010-March/thread.html Best A. Mani -- A. Mani ASL, CLC, AMS, CMS http://www.logicamani.co.cc _______________________________________________ network mailing list [email protected] http://lists.fosscom.in/listinfo.cgi/network-fosscom.in
