Hi Wayne, I shrink the Audacity window to a minimum, so it ain't no problem pressing record in due time and cut the beginning second w/o audio signal afterwords. Rotter works automatically according to its internal time schedule (every hour a new file) as far as I understand. In addition it produces .mp3 in 128 kBit only - witch is way to poor for professional FM broadcast quality. It is a nice tool for documentation purposes though.
Cheers, Chris. On Sun, 5 Aug 2012 14:50:47 +0100 "Wayne Merricks" <[email protected]> wrote: > Lots of good info here, regarding Audacity for recording. Would something > like rotter be easier considering you could automate it? Having to click > record in Audacity seems a bit clunky to me or is there another mysterious > way of using Audacity that I don't know about? > > > -----Original Message----- > From: [email protected] on behalf of Chris Cramer > Sent: Sun 05/08/2012 13:39 > To: [email protected] > Subject: Re: [RDD] RMS levels (some definitions) > > Hi all, > > About levels, > > this is a part of the mastering process of a recording and there might be a > reason why this is still done manually in the CD production process. > Never the less as fas as I know there is only one product in the market that > is able to calculate and display a volume level of an audio signal. > That would be the Peak Program Meters (PPM) from RTW. And that is only a > display, not an algorithm for sound processing. > This does not exist yet (as far as I know) as the material that is supposed > to be processed might be of totally different dynamic nature. > A voice track has an other dynamic range than a classical music track or a > techno club track or a rock ballade for example. > Therefore it is nearly impossible to pre program a one-fits-all algorithm. > In my studio I do have a Jünger Audio digital dynamic processor that I use > for vinyl copies or raw audio material that was recorded live that has to be > processed. > In my cart library I process manually watching my external ppm and USING MY > EARS to find a matching level. > As I mainly use RIVENDELL for pre production I process the final show using > the JACK plugin JAMIN witch performs very good (without any pumping) to > produce a -0.2 dBFS audio stream I record using Audacity at the same time. > When finished I export the recording as .wav and process it with lame in high > quality. This file is then uploaded to the dropbox of the broadcast computer > and then aired as scheduled. > > About working levels > I hear different opinions about levels in this group. > > There are clear definitions about levels in a professional broadcast > environment. > > First: 0 dBFS means the maximum level w/o distortion in a digital environment > (FS = Full Scale) > > In the area of the European Broadcast Union (EBU) the following levels have > been agreed on: > > Nominal Level and Test Tones: > +6 dBU = 1,550 V = 0 dBr (VU) = -9 dBFS > > In the area of the Audio Engineers Society (AES) the following levels have > been agreed on: > > Nominal Levels and Text Tones: > +4 dBU = 1.228 V = 0 VU = -20 dBFS > > Why? > > EBU > +6 dBU was selected to produce a high signal/noise radio in a symmetric line > environment > -9 dBFS was selected because large digital headrooms are not a necessity in a > pre processed audio signal environment > 0 dBr is the 0 dB mark on a PPM > > AES > -20 dBFS was selected to provide enough digital headroom in a live signal > environment in order to protect the live recorded material from clipping in a > digital environment > > CD / DVD production > In the beginning of the digital audio age a CD was produced AAD (Analogue > Recording, Analogue Mastering, Digital Product): > The recording was made on a analogue multitrack recorder such as STUDER and > then mixed down in a studio on a 2 track tape (mainly with DOLBY SR or TELCOM > C noise reduction). > This tape was then processed in a PREMASTERING STUDIO. There this tape was > EQed and dynamically processed and then recorded on a U-MATIC digital Audio > Recorder with pq encoding. > The pq encoding was the track, subtrack and pause marks as well as the index > (Table Of Contents, TOC) of the CD. > As there was NO digital audio processing at that time it was a lot of work to > copy the analogue tape as the individual peaks had to be found out first in > oder to provide the maximum available dynamic range for the recording. > In addition there is an option called emphasis - this is some sort of noise > reduction in a digital environment. If you copy a CD digitally there might be > a change in the treble. That is caused by emphasis. The track would need > deemphasis. > Today digital audio processing is the daily business in the recording > industry and therefore the recordings appear much louder. The typical CD > shows a level of -0.2 dBFS. Theoretically 0 dBFS would be possible and some > unprofessional mastering guys provide premasters like that to the > manufacturing plants. But it makes sense to keep masters at -0.2 dBFS to > ensure there is no digital clipping. Some CD players actually cannot handle 0 > dbFS and produce clipping during playback. In addition a prolonged 0 dBFS is > considered a digital clip as it is unknown weather this really is a clipping > of a signal that normally would extend above the 0 dBFS or not... > > How to measure levels > A classical VU meter is not aligned to integration times - therefore it is > not suitable for a professional level measurement. > To measure a line audio level an integration time of 10ms has internationally > been agreed on > To measure a digital audio level the peak sample is what counts. So there is > no integration time, the measurement time frame equals the sampling rate. > For the fallback time a value of 1.7s (+/- 0.3s) / 20 dB is acceptable > The display range according to DIN 45406 / EBU / IEC 268-10 should be -50dB > to +9dB if used in an EBU environment > It makes sense to provide a peak hold function and to use at least 200 > segments for accurate readability. > RTW and other companies use different brightness values or additional bars to > display both the analogue and the digital integration time measurement > results and (in case of RTW) the calculated loudness at the same time. > However it appears to be a problem for most audio applications to provide an > accurate level display in their applications. > Maybe a programmer would like to implement the above values into the > RIVENDELL working environment. I would love it! In addition it would be GREAT > if the user would be able to adjust the system level of RIVENDELL according > to its working environment display wise. I am located in the EBU area and I > work with -9 dBFS for 0 dBr (VU). So sadly the built in Rivendell level > meters will always display an incorrect level. > > Cheers, > Chris. _______________________________________________ Rivendell-dev mailing list [email protected] http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
