Right. I forgot: rotter suports higher bitrates but no high quality encoding...
On Sun, 5 Aug 2012 16:15:32 +0200 Chris Cramer <ch...@smartvia.de> wrote: > 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" <waynemerri...@thevoiceasia.com> 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: rivendell-dev-boun...@lists.rivendellaudio.org on behalf of Chris > > Cramer > > Sent: Sun 05/08/2012 13:39 > > To: rivendell-dev@lists.rivendellaudio.org > > 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 > Rivendell-dev@lists.rivendellaudio.org > http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev _______________________________________________ Rivendell-dev mailing list Rivendell-dev@lists.rivendellaudio.org http://lists.rivendellaudio.org/mailman/listinfo/rivendell-dev
