-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 rotter's mp3 encoder uses LAME, widely regarded as the absolute best MP3 encoder out there. At 320k (-f mp3 -b 320), it's perfectly usable. 'In the EBU area' is quite a broad stroke - at our community radio station, >192k MP3 was accepted only as a last resort, with FLAC/WAV being the preferred format for ingest.
I also had a tiny little bash script that, every hour, split the last hour's files into two half-hour files and reencoded from FLAC to MP3 for the purposes of helping presenters produce podcasts. rotter's a great tool in widespread use, not just for documentation purposes... Cheers, James Harrison On 05/08/2012 16:31, Chris Cramer wrote: > Hi James, > > as I said, a really nice tool for documentation purposes. > the most applications that support .mp3 do not provide an option for high quality encoding. > In the EBU area 320kBit in high quality (yes! BIG difference!) is considered broadcast quality. > That is why I produce .wav files and encode manually... > > The main topic was about levels though - that actually was my main issue :-) > > Cheers, > Chris > > On Sun, 05 Aug 2012 16:19:59 +0100 > James Harrison <ja...@talkunafraid.co.uk> wrote: > > rotter supports FLAC, WAV, MP3, Vorbis. I've been using it in > production for lossless (4-week) archives and lossy (6 month) archives > for over a year and a half now. It's stable and works flawlessly. > > Cheers, > James Harrison > > On 05 August 2012 15:27:38, Chris Cramer wrote: > >>> 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 >> >> _______________________________________________ >> 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 -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.17 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlAelFUACgkQ22kkGnnJQAwPyACfeK7kpP1zllt8aIbEC5SFpMEY 3EUAoK3V9Ny7w6ZIGObhUfAbsyLwS7+J =YcTi -----END PGP SIGNATURE-----
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