Home

Maintaining Audio Quality in the Broadcast/Netcast Facility

1. In digital distribution systems 20 bit words 120dB dynamic range are usually adequate to represent the signal accurately 20 bits can retain the full quality of a 16 bit source even after as much as 24dB attenuation by a mixer There are almost no A D converters that can achieve more than 20 bits of real accuracy and many 24 bit converters have accuracy considerably below the 20 bit level Marketing bits in A D converters are outrageously abused to deceive customers and if these A D converters were consumer products these bogus claims would be actionable by the Federal Trade Commission There is considerable disagreement about the audible benefits if any of raising the sample rate above 44 1 kHz An extensive double blind test using 554 trials showed that inserting a CD quality A D A loop into the output of a high resolution SACD player was undetectable at normal to loud listening levels by any of the sub jects on any of four playback systems The noise of the CD quality loop was audible only at very elevated levels Moreover there has been at least one rigorous test comparing 48 kHz and 96 kHz sample rates This test concluded that there is no audible difference between these two sample rates if the 48 kHz rate s anti aliasing filter is appropriately designed 11 Meyer E Brad Moran David R Audibility of a CD Standard A DA A Loop Inserted into High Resolution Audio Playback JAES Volume 55 Issue 9 pp
2. carrier and the peak limiting process pro duces modulation sidebands around each Fourier component 53 Maintaining Audio Quality Considered this way a hard clipper has a wideband gain control signal and thus in troduces sidebands that are far removed in frequency from their associated Fourier carriers Hence the carriers have little ability to mask the resulting sidebands psychoacoustically Conversely a look ahead limiter s gain control signal has a much lower bandwidth than that of a clipper and produces modulation sidebands that are less likely to be audible Simple wideband look ahead limiting can still produce audible intermodulation dis tortion between heavy bass and midrange material The look ahead limiter algo rithm in Optimods uses sophisticated techniques to reduce such IM distortion with out compromising loudness capability Conventional AM FM or TV audio processors that employ pre emphasis de emphasis and or clipping peak limiters do not work well with perceptual audio cod ers such as AAC HE AACv1 v2 The pre emphasis de emphasis limiting in these proc essors unnecessarily limits high frequency headroom Further their clipping limiters create high frequency components distortion that the perceptual audio coders would otherwise not encode In addition several audio processing manufacturers offer pre processing claimed to minimize codec artifacts at low bitrates Orban s technology is call
3. Sticky Shed Syndrome Tape manufactured from the 1970s through the 1990s particularly by AGFA Am pex and 3M may suffer from so called sticky shed syndrome When played the tape sticks to the guides of the playback machine and severe oxide loss may occur The generally accepted cure is to bake the tape at 130 F 54 C in a convection oven One recommended device is the Snackmaster Pro model FD 50 made by American Harvest2 Don t use the oven in a household stove or a microwave oven Baking time ranges from about 4 hours for 1 4 tape to 8 hours for 2 tape although it s not critical You can t over bake unless you leave the tape in for a day or so if you under bake and the tape is still gummy you can bake it more After you shut off the heat leave the tape to cool down to room temperature before attempting to play it A baked tape should be playable for about a month although this depends greatly on the ambient humidity Although many tapes can be re baked as necessary this is not always true baking has risks2 It is important to make a high quality digital ar chive of the tape on its first pass through the playback machine after baking This 22 800 288 4545 www americanharvest com Model FD 50 is no longer being manufactured However American Harvest still makes similar products which we have not evaluated 23 Bill Holland Industry s Catalog at Risk Archived Tapes could be Lost to Binder Problem Bi
4. not accurately compensate for the shape of the rolloff caused by a worn head 6 Record and maintain alignment properly Alignment tapes wear out With wear the output at 15 kHz may be reduced by several dB If you have many tape machines to maintain it is usually more eco nomical to make your own secondary standard alignment tapes and use these for weekly maintenance while reserving your standard alignment tape for refer ence use See below However a secondary standard tape is not suitable for critical azimuth adjustments These should be made using the methods described above employing a test tape recorded with a full track head Even if you happen to have an old full track mono machine getting the azimuth exactly right is not practical use a standard commercial alignment tape for azimuth adjustments The level accuracy of your secondary standard tape will deteriorate with use check it frequently against your primary standard reference tape Because ordi nary wear does not affect the azimuth properties of the alignment tape it should have a very long life if properly stored Store all test tapes e Tails out e Under controlled tension e In an environment with controlled temperature and humidity 66 Maintaining Audio Quality e With neither edge of the tape touching the sides of the reel this can only be achieved if the tape is wound onto the storage reel at normal play back record speeds and not at
5. Transmitters Proc 1987 Broadcast Engineering Conf National Assoc of Broadcasters Las Vegas NV pp 43 52 Available from NAB Member Services in the Broadcast Netcast Facility AM Antenna The benefits of a transmitter with a digital modulator will only be appreciated if it feeds an antenna with wideband symmetrical impedance A narrowband antenna not only audibly reduces the high frequency response heard at the receiver but also can cause non linear distortion in radios envelope detectors if asymmetrical imped ance has caused the upper and lower sidebands to become asymmetrical Such an tennas will not work for any of the AM IBOC systems proposed at this writing Correcting antennas with these problems is specialized work usually requiring the services of a competent consulting engineer DAB HD Radio Netcasting Encoders Most often netcasts and podcasts use lossy compression at bitrates below 64 kbps At these bitrates audio quality depends critically on the choice of audio codec At this writing the highest quality codec at bitrates of 24 to 64 kbps codec is HE AAC v2 Refer to Data Compression on page 12 for a detailed discussion of transmission codecs Be aware that not all codec implementations sound the same Even though various implementations of a specific codec type may encode decode audio in the same for mat the various implementations may not all produce the same audio quality These codecs are only as goo
6. and will usually suggest ways by which relatively inexpensive acoustic treatment absorption and diffusion can improve room acoustics With the advent of low cost personal computers and sound cards it is possible to buy economical software to do room analysis and tuning Since the invention of TDS a number of other techniques like MLSSA Maximum Length Sequence System Analyzer http mlssa com have been developed for measuring and tuning rooms with accuracy greater than that provided by traditional third octave analyzers It is certainly true that room acoustics must be optimized as far as economically and physically possible before electronic equalization is applied to the monitor system If room acoustics and the monitor are good equalization may not be necessary Once room acoustic problems have been solved to whatever extent practical make frequency response measurements to determine what equalization is required A MLSSA analyzer a TDS analyzer dual channel FFT analyzer or a third octave ana lyzer can be used for the measurements To obtain meaningful results from the ana lyzer the calibrated microphone that comes with the analyzer should be placed 45 46 Maintaining Audio Quality where the production engineer s ears would ordinarily be located If a third octave analyzer is used excite each loudspeaker in turn with pink noise while observing the acoustic response on the analyzer If a MLSSA or TDS analyzer is use
7. high frequency response or align the master recorder by ear Adjust for the crispest sound while listening to the mono sum of the announcer s voice on the standard alignment tape the azimuth on the announcer s voice will be just as accurate as the rest of the tape If the traditional Lissajous pattern is used use several frequencies and adjust for minimum differential phase at all frequencies Using just one frequency 15 kHz for example can give incorrect results 8 Check record alignment and adjust as necessary Set record head azimuth bias equalization and calibrate meters according to the manufacturer s recommendations We recommend that tape recorders be ad justed so that 4dBu or your station s standard operating level in and out cor responds to OVU on the tape recorder s meters to Dolby level and to standard operating level This is ordinarily 250 nW m for conventional tape and 315 nW m for high output tape refer to the tape manufacturer s specifications for rec ommended operating fluxivity in the Broadcast Netcast Facility Current practice calls for adjusting bias with the high frequency overbias method rather than with the prior standard peak bias with 1 5 mil wave length method To do this record a 1 5 mil wavelength on tape 5 kHz at 7 5ips and increase the bias until the maximum output is obtained from this tape Then further increase the bias until the output has decreased by a fi
8. the added data in an attempt to render it inaudible To maximize the data throughput the average level of the program audio should be consistently high This maximizes the ability of the PPM encoder to inject its data while ensuring that the program audio masks the data While a simple AGC will help compared to no audio processing at all a full audio processing chain including an AGC multiband compressor and peak limiting will work significantly better than an AGC alone Because the PPM signal amplitude is very low with respect to the program audio the PPM signal can in principle be added to the final peak limited audio signal without significantly disturbing peak modulation and without compromising loud ness However there are potential pitfalls For digital broadcasting or netcasting adding the PPM signal is easily done by inserting the Arbitron PPM encoder after the peak limited signal just before the broadcast netcast digital audio encoder in put This is best done in the digital domain by using an Arbitron PPM digital en coder which does not compromise the program audio waveform fidelity and which has sufficient headroom If an Arbitron PPM analog encoder is used one must to pay careful attention to headroom to prevent the encoder from clipping the audio Furthermore it is necessary to determine if the analog encoder signal path intro duces overshoot and or tilt into the processed audio If it does so it can cause peak clipping t
9. which also tries to minimize CD damage caused by careless handling places each CD in a protective plastic caddy The importance of handling CDs with care and keep ing the playing surface clean cannot be over emphasized Contrary to initial market ing claims of invulnerability CDs have proven to require handling comparable to that used with vinyl disks in order to avoid broadcast disasters Except for those few CD players specifically designed for professional applications CD players usually have unbalanced 10dBV outputs In many cases it is possible to interface such outputs directly to the console by trimming input gains without RFI or ground loop problems To solve any problems several manufacturers produce low cost 10dBV to 4dBu adapters for raising the output level of a CD player to pro fessional standard levels A peak overload issue commonly called 0 dBFS can be another problem with the analog outputs of CD players and computer sound cards If the DAC reconstruction filter does not have 3dB of headroom above 0 dBFS the DAC is likely to clip the au dio coming from today s aggressively hyper processed CDs This will add even more distortion to the regrettable amounts of clipping distortion that are already intro duced in the CD mastering process unless oversampled limiting which anticipates and compensates for 0 dBFS overshoots was used when the CD was mastered However many of today s CDs are hard clipped in th
10. which often improves the quality However beware of remixes so radical that they no longer sound like the hit version as remembered by audiences Furthermore many of these original performances may be available on many different albums possibly from the same label but from different sources with varying quality A valu able on line resource for this information is Top 40 Music on Compact Disc http www top40musiconcd com Some of the more recent remasterings may contain additional signal processing be yond simple click and pop elimination Newly remastered tracks should be validated very carefully as the newer tracks may suffer from excessive digital limiting that re duces transient impact and punch Therefore the older less processed sources may stand up better to Optimod transmission processing Some computer audio editing software such as Adobe Audition Steinberg WaveLab iZotope RX and Diamond Cut DC8 contains restoration tools that attempt to undo clipping This is a challeng ing task so these tools should be qualified and used very carefully One size does not fit all Unfortunately there is no reliable formula for choosing old or new CDs For exam ple some original CD releases were simply transfers of a vinyl pressing The limita tions of vinyl are usually audible and subsequent remasters were a dramatic im provement either because the original master tapes were discovered and used or because improved vinyl restoration
11. Appendix Analog Media Authors Note for the 2014 Edition This Appendix devotes considerable space to the vagaries of analog media vinyl disk and analog tape that are becoming less and less important in broadcast pro duction However given that they exist and that archival material may be stored on such media we have chosen to retain this material with minor editing in the cur rent revision Because these media are analog they require far more tweaking and tender loving care than do the digital media discussed above For this reason the following sections are long and detailed Vinyl Disk Some radio programming still comes from phonograph records either directly or through dubs Not only are some club DJs mixing directly to broadcast netcast from vinyl but also some old recordings have not been re released on CD Even if they have been a disturbing fact is that many recently remastered CDs sound far worse than the original vinyl releases This section discusses how to accurately retrieve as much information as possible from the grooves of any record Vinyl disk is capable of very high quality audio reproduction Consumer equipment manufacturers have developed high fidelity cartridges pick up arms turntables and phono preamps of the highest quality Unfortunately much of this equipment has insufficient mechanical ruggedness for the pounding that it would typically receive in day to day broadcast operations There are
12. Audio which also offers 24 bit resolution and 96 kHz sample rate was a commercially unsuccessful and now obsolete attempt to raise the quality of consumer media while SACD which uses bitstream coding DSD Direct Stream Digital instead of the CD s PCM Pulse Code Modulation is as of this writing hang ing on as a niche format primarily for classical music Most prognosticators believe that the future of audiophile quality music lies in downloadable high resolution files that use lossless compression These files are already available from several sources Because most audio is still sourced at a 44 1 kHz sample rate upsampling to 48 kHz does not improve audio quality Further many broadcast digital sources have re ceived various forms of lossy data compression While we had expected the black vinyl disk to be obsolete by this revision it is still used in specialized applications like live club style D J mixing and re mastering when the original recordings are no longer available or are in poor condition Although digital to analog conversion technology is constantly improving we be lieve that some general observations could be useful In attempting to reproduce CDs with the highest possible quality in the analog domain the industry has settled into technology using delta sigma digital to analog converters DACs with ex treme oversampling These converters use pulse width modulation or pulse duration modulation techniqu
13. The advantage to asynchronous digital audio is that just about any thing can be connected to just about anything else without difficulty However there are cautions 27 28 Maintaining Audio Quality Computer audio can confuse those trying to achieve bit accurate audio Because au dio has many different sources and destinations computer operating systems con tain software sample rate converters for both record and play to make things easier for the average user For the professional this can cause performance problems and sample rate conversion can occur when it is not wanted or expected To work around this problem most professional users use the ASIO interface and drivers However not all audio applications support ASIO Furthermore if recording or encoding from another application is required this is not supported by ASIO Microsoft Windows XP has a high quality sample rate converter in the audio stack for both record and play To achieve all of its potential quality adjust XP s Sample rate conversion quality to Best The control is located in Sound and Audio Device Properties gt Audio gt Sound Playback gt Advanced gt Performance To prevent sample rate conversion from occurring be certain that source sample rates match record sample rates and that play sample rates match destination sam ple rates If sample rate conversion is unavoidable Windows XP will provide a high quality conversion Microsoft implemented a n
14. The performance of any loudspeaker is strongly influenced by its mounting location and room acoustics If room acoustics are good the third octave real time analyzer provides an extremely useful means of measuring any frequency response problems intrinsic to the loudspeaker and of partially indicating problems due to loudspeaker placement and room acoustics By their nature the third octave measurements combine the effects of direct and reflected sound This may be misleading if room acoustics are unfavorable Problems can include severe standing waves a reverberation time which is not well behaved as a function of frequency an insufficient number of normal modes Eigen modes lack of physical symmetry and numerous problems which are discussed in more detail in books devoted to loudspeakers and loudspeaker equalization Time Delay Spectrometry TDS is a technique of measuring the loudspeaker room interface that provides much more information about acoustic problems than does the third octave real time analyzer TDS which some sound contractors are licensed to practice is primarily used for tuning recording studio control rooms and for ad justing large sound reinforcement systems The cost may be prohibitive for a small or medium sized station particularly if measurements reveal that acoustics can only be improved by major modifications to the room However TDS measurements are highly useful in determining if LEDE criteria are met
15. Windows based on sample rate If a record encode application operates at a differ ent sample rate than that listed by the audio device either Windows Core Audio or the application must do sample rate conversion If Win dows Core Audio provides the sample rate conversion the audio signal is subject to the performance of that conversion algorithm It is therefore very important that this conversion be performed as cleanly as possible in the Broadcast Netcast Facility 2 9 Anyone working with computer digital audio is urged to update all Win dows 7 Server 2008 computers with this Hotfix Metering One of the most misunderstood details of audio is exactly how to measure levels and how analog levels relate to digital levels The diagram below shows the calibra tion and level relationships between the following meters where all meters are dis playing a sinewave at SMPTE reference level e True Peak Reading Digital Level Meter e VU Meter ANSI e Peak Program Meter EBU IEC 268 10 IIB e Peak Program Meter UK IEC 268 10 IIA e Peak Program Meter Nordic IEC 268 10 I e Peak Program Meter DIN IEC 268 10 DIN 45406 Reference 20 dBFS 0 dBFS HEADROOM 60 50 45 40 35 30 25 20 15 10 8 6 4 2 0 30 Maintaining Audio Quality Measuring and Controlling Loudness Orban now offers a loudness meter application for Windows Vista and 7 and for Mac It is available for
16. and or multipath Each CD that is transferred should be checked by ear to ensure that the left and right chan nels sum to mono without artifacts CDs that sound fine in stereo may suffer from high frequency loss or flanging caused by uncorrected relative time delays be tween the left and right channels Some computer audio editing software such as Adobe Audition contains restoration tools like Automatic Phase Correction With careful adjustment possibly even in manual mode good results are achievable Some CDs sourced from full track mono masters have been transferred using a ste reo playback tape head The slight delay and azimuth differences between the two gaps can cause high frequency loss when summed to mono which can be com pletely eliminated by choosing the best channel from the CD and using it as a source for both channels of the file in your playout system Some music was poorly recorded and mastered There are many reasons for this in cluding poor equipment or monitoring environments Some cases will require care ful equalization to achieve consistent results Maintaining Audio Quality Many tracks even from desirable labels have been recently re mastered and may sound quite different from the original transfer to CD Regardless of source it is wise to use the original performance even if its audio quality is worse than alterna tive versions Sometimes the original performance has been remixed for CD release
17. be absolutely perpendicular to the disc to sustain a good separation To prevent a fixed tracking error the cartridge must be parallel to the headshell Overhang should be set as accurately as possi ble 1 16 inch 0 16 cm and the vertical tracking angle should be set at 20 by adjusting arm height A valuable tool for precision alignment is the Protractor NG available from Dr Feickert Analogue http www feickert de engl protractor html 2 Adjust the tracking force correctly Usually better sound results from tracking close to the maximum force recom mended by the cartridge manufacturer If the cartridge has a built in brush do not forget to compensate for it by adding more tracking force according to the manufacturer s recommendations Note that brushes usually make it impossible to back cue although this should not be done when transferring to digital anyway 3 Adjust the anti skating force correctly The accuracy of the anti skating force calibration on many pick up arms is ques tionable The best way to adjust anti skating force is to obtain a test record with an extremely high level lateral cut some IM test records are suitable Connect the left channel output of the turntable preamp to the horizontal input of an oscilloscope and the fight channel preamp output to the vertical input Operate 57 58 Maintaining Audio Quality the scope in the X Y mode such that a straight line at a 45 degree angle i
18. broadcast equipment is necessary in the production studio because quality loss here will repeatedly ap pear on the air The production director should be acutely sensitized to audible quality degradation and should immediately inform the engineering staff of any problems detected by ear Minimize motor noise To prevent motor noise from leaking into the production microphone equip ment with noisy fans and hard drives should be installed outside the studio if possible Otherwise they should reside in alcoves under soffits surrounded by acoustic treatment In the real world of budget limitations this is sometimes not possible although sound deadening treatment of small spaces is so inexpensive that there is little excuse for not doing it But even in an untreated room it is possible to use a directional microphone with figure eight configuration for ex ample with the noisy machine placed on the microphone s dead axis Choos ing the frequency response of the microphone to avoid exaggerating low fre quencies will help In particularly difficult cases a noise gate or expander can be used after the microphone preamp to shut off the microphone except during ac tual speech Consider processing the microphone signal Audio processing can be applied to the microphone channel to give the sound more punch Suitable equalization may include gentle low and high frequency boosts to crispen the sound aid intelligibility and add a b
19. common problem is inadequate record amplifier head room In many cases it is possible to improve the situation by increasing the operating current in the final record head driver transistor to a value close to its power dissipation limits This is usually done by decreasing the value of emitter and sometimes collector resistors while observing the collector voltage to make sure that it stays at roughly half the power sup ply voltage under quiescent conditions and adjusting the bias network as necessary if it does not About the Authors Robert Orban Robert Orban received the B S E E degree from Princeton University in 1967 and the M S E E degree from Stanford University in 1968 In 1970 he founded Orban Associates originally as a manufacturer of studio equip ment In 1975 Orban Associates introduced the original Optimod FM 8000 which was the first in a long line of broadcast audio processors for AM FM TV and digital broadcasting from the company Although Orban Associates changed hands several times in the ensuing decades Orban continues to work at the successor company as Vice President Chief Engineer Starting at age six he began studying piano and voice A knack for improvisation and musical composition led him at age 15 to create a setting of Psalm 108 for mixed chorus and piano which was performed publicly to considerable acclaim by the chorus at his high school in Butler New Jersey At Princeton in addition to the
20. for example to keep DC offsets out of faders to prevent scratchiness If capacitors must be used polystyrene polypropylene or polycar bonate film capacitors are preferred If electrolytic capacitors are used it is wise to use them with DC bias so that AC audio signals can never reverse bias them To eliminate DC offsets the best audio designs use servos instead of coupling ca pacitors However if it is impractical to eliminate capacitors or to change capacitor types do not be too concerned it is probable that other quality limiting factors will mask the capacitor induced degradations Of course the number of transformers in the audio path should be kept to an ab solute minimum However transformers are sometimes the only practical way to break ground loops and or eliminate RFI If a transformer is necessary use a high quality device like those manufactured by Jensen or Lundahl 9 Jensen Transformer Inc Chatsworth California USA Phone 1 818 374 5857 or Fax 1 818 374 5856 10 Lundahl Transformers AB Tibeliusgatan 7 SE 761 50 Norrt lje SWEDEN Phone 46 176 139 30 Fax 46 176 139 35 35 36 Maintaining Audio Quality In summary the path to highest analog quality is that which is closest to a straight wire More is not better every device removed from the audio path will yield an im provement in clarity transparency and fidelity Use only the minimum number of amplifiers capacitors and trans
21. ground loops which induce hum and can cause oscillation In an ideal system e All units in the system should have balanced inputs In a modern system with low output impedances and high input impedances a balanced input will provide common mode rejection and prevent ground loops regardless of whether it is driven from a balanced or unbalanced source e All equipment circuit grounds must be connected to each other all equipment chassis grounds must be connected together e Ina low RF field cable shields should be connected at one end only preferably the destination input end This also prevents input noise pick up when the out put is disconnected e Ina high RF field audio cable shields should be connected to a solid earth ground at both ends to achieve best shielding against RFI e Whenever coaxial cable is used shields are automatically grounded at both ends through the terminating BNC connectors Audio over Ethernet Audio over Ethernet is available as Layer 2 and Layer 3 protocols Layer 2 is not TCP UDP and cannot be routed with common IP routers The most common Layer 2 protocol is ABV whose main advantage is very low latency The Audio Engineering Society has published to AES 67 standard which specifies interoperability requirements for Layer 3 connections These connections use UDP RTP multicast for audio packets and TCP RTSP for stream initialization They use PTP clock for synchronization At this writing RAVE
22. is not possible to maintain unity gain through the volume control Microsoft Windows Media Player and Adobe Flash Players should be operated at 100 O dBFS at all times and level control should be done either at the amplifier volume control or console fader In the absence of noise shaping the spectrum of the usual triangular probability function TPF dither is white that is each arithmetic frequency incre ment contains the same energy However noise shaping can change this noise spec trum to concentrate most of the dither energy into the frequency range where the ear is least sensitive In practice this means reducing the energy around 4 kHz and raising it above 9 kHz Doing this can increase the effective resolution of a 16 bit 37 38 Maintaining Audio Quality system to almost 19 bits in the crucial midrange area and is standard in CD master ing There are many proprietary curves used by various manufacturers for noise shaping and each has a slightly different sound It has been shown that passing noise shaped dither through most classes of signal processing and or a D A converter with non monotonic behavior will destroy the advantages of the noise shaping by filling in the frequency areas where the origi nal noise shaped signal had little energy The result is usually poorer than if no noise shaping had been used For this reason Orban has adopted a conservative approach to noise shaping recommendin
23. is progressively less argument for storing programming using lossy compression and contribution codecs have thus fallen out of favor Of course either no compression or lossless compression will achieve the highest quality There is no quality difference between these Uncompressed audio workflow is much easier to deal with and prevents the inexorable quality loss inherent in the codec de code encode cycle in the Broadcast Netcast Facility Conversely cascading stages of lossy compression can cause noise and distortion to become unmasked Multiband audio processing can also cause noise and distortion to become unmasked because multiband processing automatically re equalizes the program material so that the frequency balance is not the same as the fre quency balance seen by the psychoacoustic model in the encoder Storing audio in linear PCM format makes the audio easier to edit and copy without quality loss Sony s MiniDisk format is a technology that combines data compression Sony A TRAC and random access disk storage While not offering the same level of audio quality as CD R or CD RW these disks are useful for field acquisition or other appli cations where open reel or cassette tape had been previously used They offer nota bly higher quality than the analog media they replace along with random access and convenient editing However recorders using flash RAM have effectively obso leted them Many facilities are receiving
24. list of compatible writers is available on the M DISC website Digital Tape While DAT was originally designed as a consumer format it achieved substantial penetration into the broadcast environment This 16 bit 48 kHz format is theoreti cally capable of slightly higher quality than CD because of the higher sample rate In the DAR environment where 48 kHz sample rate is typical this improvement can be 1 http www mdisc com in the Broadcast Netcast Facility 1 1 passed to the consumer However because the sample rate of the FM stereo sys tem is 38 kHz there is no benefit to the higher sampling rate by the time the sound is aired on FM The usual broadcast requirements for ruggedness reliability and quick cueing apply to most digital tape applications and these requirements proved to be quite diffi cult to meet in practice The DAT format packs information on the tape far more tightly than do analog formats This produces a proportional decrease in the dura bility of the data To complicate matters complete muting of the signal rather than a momentary loss of level or high frequency content as in the case of analog ac companies a major digital dropout At this writing there is still debate over the reliability and longevity of the tape Some testers have reported deterioration after as little as 10 passes while others have demonstrated almost 1000 passes without problems Each demonstration of a tape surviving hun
25. only two reasonably high quality cartridge lines currently made in the USA that are generally accepted to be sufficiently durable for professional use the Stanton and the Shure professional series Although rugged and reliable these car tridges do not have the clean transparent operation of the best high fidelity car tridges This phono cartridge dilemma is the prime argument for transferring all vi nyl disk material to digital media in the production studio and broadcasting only from digital media In this way it is possible with care to use state of the art car tridges arms and turntables in the dubbing process which should not require the mechanical ruggedness needed for broadcast equipment Good high quality turntables and tonearms have become a bit scarce However the Technics SP 10 and its associated base SH 10B3 and tonearm EPA B500 EPA A250 EPA A500 are very good choices for mastering vinyl to digital This reduces the problem of record wear as well Production facilities specializing in high quality transfer of vinyl to digital media should consider supplementing their conventional turntable with an ELP Laser Turn table Instead of playing disks mechanically this pricey device plays vinyl without mechanical contact to the disk using laser beams instead The authors have thor 16 http www elpj com in the Broadcast Netcast Facility oughly evaluated the ELP and we can recommend it as delivering higher aud
26. second and higher Over the years there have been many MP3 encoder implementations having widely varying audio quality Because of this it is undesirable to use Layer 3 MP3 files for playout system sources if high audio quality is an objective Because Layer 2 can be audibly transparent at high bitrates it is suitable for both contribution and transmission although its low coding efficiency compared to AAC Advanced Audio Coding make it obsolete for transmission Layer 2 audio can be easily identified by viewing using an FFT spectrum analyzer The upper bands or bins will appear to gate on and off as the audio plays AAC HE AAC MPEG 2 Advanced Audio Coding was designed for use as a transmission codec as the successor to MPEG 1 Layer 3 It incorporates numerous improvements to the Layer 3 algorithm including significantly improved pre echo performance Blind tests show that AAC demonstrates greater sound quality and transparency than MP3 for files coded at the same bitrate AAC HE AAC codec technology combines three MPEG technologies AAC Coding Technologies Spectral Band Replication SBR and Parametric Stereo PS SBR is a bandwidth extension technique that enables audio codecs to deliver similar quality at half the bitrate of codecs that do not use SBR Parametric Stereo increases the codec efficiency a second time for low bitrate stereo signals SBR and PS are forward and backward compatible methods to enhance the effi cienc
27. software and techniques were employed On the other hand many remasters were subject to additional dynamic range compression and peak limiting and do not sound as good as their original releases even though the newer remasters may claim higher resolution We believe that the best sounding CDs are probably those mastered from about 1990 to 1995 Before 1990 many mastering engineers used the Sony PCM1610 s con verters because the standard medium for transmitting a mastered CD to a replica tion house was then a 3 4 U Matic video recorder and the Sony formatted the audio so that it could be recorded as video on such a machine However the PCM1610 s converters were widely criticized for their sound There were too many low slewrate opamps and electrolytic capacitors in the signal path not to mention high order analog anti aliasing filters Things changed around 1990 with the introduction of analog to digital converters based on the then new Analog Devices AD1879 delta sigma A D converter chip This chip uses oversampling to eliminate the need for high order analog anti aliasing filters and has very good low level linearity The resulting signal path has very little group delay distortion If properly designed converters based on the AD1879 sound very good Meanwhile the loudness wars were still at least five years in the future Starting around 1995 average levels on CDs started to increase The availability of early digital look ahead lim
28. source material that has been previously processed through a lossy data reduction algorithm whether from satellite over landlines or over the Internet Sometimes several encode decode cycles will be cascaded before the material is finally broadcast As stated above all such algorithms operate by in creasing the quantization noise in discrete frequency bands If not psychoacousti cally masked by the program material this noise may be perceived as distortion gurgling or other interference Cascading several stages of such processing can raise the added quantization noise above the threshold of masking such that it is heard In addition at least one other mechanism can cause the noise to become audible at the receiver The multiband limiter in a broadcast station s transmission processor performs an automatic equalization function that can radically change the fre quency balance of the program This can cause noise that would otherwise have been masked to become unmasked because the psychoacoustic masking conditions under which the masking thresholds were originally computed have changed Accordingly if you use lossy data reduction in the studio you should use the highest data rate possible along with a codec designed for contribution like MPEG1 Layer2 at 384 kb sec or Dolby E This maximizes the headroom between the added noise and the threshold where it will be heard In addition you should minimize the num ber of encode
29. synthesizers compressors paramet ric equalizers enhancers and de essers under both the Orban and dbx brand names Orban has been actively involved in NRSC committee AM improvement work He has been widely published in both the trade and refereed press including J Audio En gineering Soc Proc Soc Automotive Engineers and J SMPTE He co authored the chapter on Transmission Audio Processing in the NAB Engineering Handbook 11 edition He currently holds over 25 U S patents In 1973 he was elected a Fellow of the Audio Engineering Society In 1993 he shared with Dolby Laboratories a Scientific and Engineering Award from the Acad emy of Motion Picture Arts and Sciences In 1995 he received the NAB Radio Engi neering Achievement Award In 2002 he received the Innovator award from Radio Magazine Today he continues to actively research new DSP audio processing tech nology and to write produce and record music Greg Ogonowski Having studied piano as a child Greg began to appreciate the sound quality and musicality of various sources It was his ear candy and his infatuation with commer cial Top 40 radio would eventually help shape and mold the way radio audio proc essing sounds today It was the summer of 1967 A Los Angeles based group called the Fifth Dimension had a hit called Up Up and Away Greg noticed it had a fresh new sound thanks to engineer Bones Howe and Greg managed to catch it playing on CKLW
30. that is unconditionally satisfying to golden eared audiophiles These limitations must be considered when discussing the quality requirements for FM electronics The problems in analog disc and tape reproduction discussed in the Ap pendix to this document are much more severe by comparison and the subtle mask ing of basic FM transmission limitations is irrelevant to those discussions AM quality at the typical receiver is far worse and golden ear considerations are completely irrelevant because they will be masked by the limitations of the receivers and by at mospheric and man made noise The four FM quality limitations are these A Multipath distortion In most locations a certain amount of multipath is un avoidable and this is exacerbated by the inability of many apartment dwellers to use rotor mounted directional antennas B The FM stereo multiplex system has a sample rate of 38 kHz so its band width is theoretically limited to 19 kHz and practically limited by the charac teristics of real world filters to between 15 and 17 kHz C Limited IF bandwidth is necessary in receivers to eliminate adjacent and al ternate channel interference Its effect can be clearly heard by using a tuner with switch selectable IF bandwidth Most stations cannot be received in wide mode because of interference But if the station is reasonably clean well within the practical limitations of current broadcast practice and
31. the suggestions above are exactly that just suggestions For a given VU or PPM indication the loudness of different talkers and different music may vary significantly A short term loudness meter like the Jones amp Torick meter can help operators maintain appropriate voice music balance by estimating more accurately than a PPM or VU the actual loudness of each program segment Many of Orban s Optimod audio processors have automatic speech music detection and can automatically change processing parameters for speech and music Setting these parameters to achieve your organization s desired speech music balance pro vides an effective way of controlling this balance automatically 33 34 Maintaining Audio Quality Electronic Quality Assuming that the transmission does not use excessive lossy compression digital au dio broadcasting and netcasting have the potential for transmitting the highest sub jective quality to the consumer and require the most care in maintaining audio qual ity in the transmission plant This is because these transmission media do not use pre emphasis and have a high signal to noise ratio that is essentially unaffected by reception conditions The benefits of an all digital facility using minimal or no lossy compression prior to transmission will be most appreciated in DAB HD SAT Radio and netcasting services FM has four fundamental limitations that prevent it from ever becoming a transmis sion medium
32. to represent the overall perceived loudness of the signal 4 Calibration with reference level A suitable average replay level is 83dB SPL A calibration relating the energy of a digital signal to the real world replay level has been defined by the SMPTE Using this calibration we subtract the current signal from the desired calibrated level to give the difference We store this difference in the audio file 5 Replay Gain The calibration level of 83dB can be added to the difference from the previous cal culation to yield the actual Replay Gain NOTE we store the differential NOT the actual Replay Gain Speech Music Balance The VU meter is very deceptive when indicating the balance between speech and music The most artistically pleasing balance between speech and music is usually achieved when speech is peaked 4 6dB lower than music on the console VU meter If heavy processing is used the difference between the speech and music levels may have to be increased Following this practice will also help reduce the possibility of clipping speech which is much more sensitive to clipping distortion than is most mu sic If a PPM is used speech and music should be peaked at roughly the same level However please note that what constitutes a correct artistic balance is highly sub jective and different listeners may disagree strongly Each broadcasting organiza tion has its own guidelines for operational practice in this area So
33. two audio channels The connection is 110Q balanced and is transformer coupled in high quality equip ment The AES3 standard specifies a maximum cable length of 100 meters While almost any balanced shielded cable will work for relatively short runs 5 meters or less longer runs require used of 110Q balanced cable like Belden 1800B 1801B plenum rated multi pair 180xF 185xF or 78xxA Single pair Category 5 5e and 6 Ethernet cable will also work well if you do not require shielding In most cases the tight balance of Category 5 5e 6 cable makes shielding unnecessary in the Broadcast Netcast Facility The AES3id standard is best for very long cable runs up to 1000 meters This speci fies 750 unbalanced coaxial cable terminated in 750 BNC connectors A 1100 7590 balun transformer is required to interface an AES3id connection to an AES3 connec tion S PDIF is a consumer digital standard closely related to the AES3 standard However S PDIF is available in two different physical interfaces coaxial and optical Coaxial is is 750 unbalanced and optical is TOSLINK Both interfaces offer excellent quality and are good for short distances Format converters are available to go between either coaxial or optical and or AES3 Grounding Very often grounding is approached in a hit or miss manner Nevertheless with care it is possible to wire an audio studio so that it provides maximum protection from power faults and is free from
34. ultra accurate equalization of the entire car tridge preamp system is most worthwhile The load capacitance and resistance should be adjusted according to the car tridge manufacturer s recommendations taking into account the capacitance of cables If a separate equalizer control is not available load capacitance and resis tance may be trimmed to obtain the flattest frequency response Failure to do this can result in frequency response errors as great as 10dB in the 10 15 kHz re gion This is very often the reason many phono cartridge evaluations often pro duce colored results The final step in adjusting the preamp is to accurately set the channel balance with a test record and to set gain such that output clipping is avoided on any re cord If you need to operate the preamp close to its maximum output level due to the system gain structure then observe the output of the preamp with an os cilloscope and play a loud passage Set the gain so that at least 6dB peak head room is left between the loudest part of the record and peak clipping in the pre amp Choose a high quality A D converter Many computer onboard or internal sound cards produce excessive noise Disable any soft clipping or limiting options in the A D Routinely and regularly replace styli One of the most significant causes of distorted sound from vinyl disk reproduc tion is a worn phono stylus Styli deteriorate sonically before any visible degrada tion can b
35. uncommon in external distribution systems The core audio input output of a computer s operating system is usually fixed point Even systems using floating point representation are vulnerable to overload at the A D converter If digital recording is used in the facility bear in mind that the over load point of digital audio recorders unlike that of their analog counterparts is abrupt and unforgiving Never let a digital recording go into the red this will almost assuredly add audible clipping distortion to the recording Similarly digital distribution using the usual AES3 connections has a very well defined clipping point digital full scale 0 dBFS and attempting to exceed this level will result in distortion that is even worse sounding than analog clipping because the clipping harmonics above one half the sampling frequency will fold around this frequency appearing as aliasing products 0 dBFS is not at all the same as 0 VU or 0 PPM In a contribution system with ade quate headroom 0 VU reference level should be placed at 20 dBFS SMPTE RP 15 standard or 18 dBFS EBU R68 standard In a transmission system where the audio will be transmitted via the Dolby AC3 codec 0 VU is often placed even lower typically 24 or 25 dBFS and the value of Dolby AC3 Dialnorm metadata transmit ted to consumers is set to match this reference level The consumer s receiver then uses the received value of Dialnorm to adjust a hidden v
36. will often be quite adequate even at low frequencies if the monitor s equal izer can be set to complement the monitor s location in the room Stereo Enhancement In contemporary broadcast netcast audio processing high value is placed on the loudness and impact of a station compared to its competition Orban originally de veloped the analog 222A Stereo Spatial Enhancer to augment a station s spatial im age achieving a more dramatic and more listenable sound The stereo image be comes magnified and intensified listeners also perceive greater loudness bright ness clarity dynamics and depth The 222A detects and enhances the attack transients present in all stereo program material while not processing other portions Because the ear relies primarily on at tack transients to determine the location of a sound source in the stereo image this technique increases the apparent width of the stereo soundstage Because only at tack transients are affected the average L R energy is not significantly increased so the 222A does not exacerbate multipath distortion While the 222A is no longer manufactured several of Orban s digital Optimods now incorporate both the 222A algorithm and a delay based algorithm in DSP Other Production Equipment The discussions in this document of disk reproduction tape digital source and elec tronic quality also apply to the production studio Uncompressed sources including CD DVD A SACD and losslessly com
37. 775 779 September 2007 12 Katz Bob Mastering Audio the art and the science Oxford Focal Press 2002 p 223 in the Broadcast Netcast Facility Assuming perfect hardware it can be shown that this debate comes down entirely to the audibility of a given anti aliasing filter design as is discussed below Never theless in a marketing driven push the record industry attempted to change the consumer standard from 44 1 kHz to a higher sampling frequency via DVD A and SACD neither of which succeeded in the mass marketplace The industry is trying again with Blu ray audio and it remains to be seen if they will be more successful than they were with DVD A or SACD Regardless of whether scientifically accurate testing eventually proves that this is audibly beneficial sampling rates higher than 44 1 kHz have no benefit in FM stereo because the sampling rate of FM stereo is 38 kHz so the signal must eventually be lowpass filtered to 17 kHz or less to prevent aliasing It is beneficial in DAB which typically has 20 kHz audio bandwidth but offers no benefit at all in AM whose bandwidth is no greater than 10 kHz in any country and is often 4 5 kHz Some A D converters have built in soft clippers that start to act when the input signal is 3 6 dB below full scale While these can be useful in mastering work they have no place in transferring previously mastered recordings like commercial CD If the soft clipper in an A D converter cannot be defe
38. 85 2009 pdf in the Broadcast Netcast Facility sound cards require input attenuation and output amplification to interface to pro fessional levels Do not use the software volume control to control input levels this cannot prevent the input A D converter from clipping Align the software output level control by setting the control as high as possible without clipping when a 0 dBFS tone file is played Many computer sound card software drivers are incompetently written and do not handle audio levels correctly To achieve professional results choose computer sound devices very carefully Several sound cards and USB audio devices have a reversed left and right audio clock that causes bit slip The resulting digital audio is not correctly time aligned which causes an interchannel phase shift that increases with frequency The sum of the left and right channels does not exhibit a flat frequency response The amount of attenuation at a given frequency depends upon the sample rate A one sample slip at 32kHz sample rate produces a notch at 16kHz and almost 6dB of loss at 10kHz 44 1kHz produces almost 3dB at 10kHz and 6dB at 15kHz 48kHz produces 2dB at 10kHz and 5dB at 15kHz Because one sample slip is audible in the mono sum devices with this problem are inappropriate for broadcast audio ap plications especially for mastering a library Many of these devices were based upon a Texas Instruments USB Codec chip that had its hardware clock re
39. EC 14496 3 2001 AMD 2 Parametric Coding for High Quality Au dio e MPEG Surround Spatial Audio Coding SAC e MPEG ISO IEC 23003 1 2007 Part 1 MPEG Surround MPEG ISO IEC 23003 2 2010 Part 2 Spatial Audio Object Coding SAOC MPEG ISO IEC 23003 2 2012 Part 3 MPEG D USAC published in early 2012 HE AACv1 is standardized by 3GPP2 3rd Generation Partnership Project 2 ISMA Internet Streaming Media Alliance DVB Digital Video Broadcasting the DVD Fo rum Digital Radio Mondiale and many others HE AACv2 is specified as the high quality audio codec in 3GPP 3rd Generation Partnership Project and all of its com ponents are part of MPEG 4 As an integral part of MPEG 4 Audio HE AAC is ideal for deployment with the H 264 AVC video codec standardized in MPEG 4 Part 10 The DVD Forum has speci fied HE AACv1 as the mandatory audio codec for the DVD Audio Compressed Audio Zone CA Zone Inside DVB H HE AACv2 is specified for the IP based delivery of content to handheld devices ARIB has specified HE AACv1 for digital broadcasting in Japan S DMB MBCo has selected HE AACv1 as the audio format for satellite mul timedia broadcasting in Korea and Japan Flavors of MPEG 4 HE AAC or its compo nents or portions thereof are also applied in national and international standards and systems such as iBiquity s HD Radio US XM Satellite Radio US and or the En hanced Versatile Disc EVD China Independent quality evaluations of AAC HE AAC
40. F351 family and the Texas Instruments TLO70 family When the highest quality is required designers will choose premium priced opamps from Analog Devices Linear Technology and Burr Brown or will use discrete class A amplifiers However the 5532 and 5534 can provide excellent performance when used properly and it is hard to justify the use of more expensive amplifiers except in specialized applications like microphone preamps active filters and composite line drivers While some designers insist that only discrete designs can provide ultimate quality the performance of the best of current ICs is so good that discrete designs are just not cost effective for broadcast netcast applications especially when the basic FM DAB audio codec quality limitations are considered Some have claimed that capacitors have a subtle but discernible effect upon sonic quality Polar capacitors such as tantalums and aluminum electrolytics behave very differently from ideal capacitors In particular their very high dissipation factor and dielectric absorption can cause significant deterioration of complex musical wave forms Ceramic capacitors have problems of similar severity Polyester film capacitors can cause a similar although less severe effect when audio is passed through them Accordingly DC coupling between stages is best and easy with opamps operated from dual positive and negative power supplies Coupling capacitors should be used only when necessary
41. G standard codecs HE AACv1 and HE AACv2 have been tested using this standard and the results have been published To our knowledge there is no published neutral third party work that assesses the Windows Media Audio codec family using the BS 1116 1 and BS 1534 1 method ologies Hence we believe that the MPEG standard codecs have more credibility MPEG1 Layer 2 3 The MPEG1 layer 2 and layer 3 codecs are the oldest technology codecs still in gen eral use today Layer 2 is a sub band audio encoder which means that compression takes place in the time domain with a low delay filter bank producing 32 frequency domain components By comparison Layer 3 is a transform audio encoder with hy brid filter bank which means that compression takes place in the frequency domain after a hybrid double transformation from the time domain Layer 2 and Layer 3 have very different ancestry and different sets of trade offs Layer 2 can offer audi bly transparent performance at high bitrates and is free from the pre echo artifacts http en wikipedia org wiki MPEG 1_Audio_Layer_ll in the Broadcast Netcast Facility that plague Layer 3 However Layer 3 s subjective performance degrades less abruptly as bitrate is reduced Layer 3 also known as MP3 was designed for use as a transmission codec and is not audibly transparent on all program material regardless of bitrate although modern implementations can sound very good at rates of 256 kb per
42. Independent tests have clearly demonstrated HE AACv2 s value In rigorous double blind listening tests conducted by 3GPP 3rd Generation Partnership Project HE AACv2 proved its superiority to its competitors even at bitrates as low as 18 kbps HE AACv2 provides extremely stable audio quality over a wide bitrate range mak ing it the first choice for all application fields in mobile music digital broadcasting and the Internet HE AACv1 has been evaluated in multiple 3rd party tests by MPEG the European Broadcasting Union and Digital Radio Mondiale HE AACv1 outperformed all other codecs in the competition 17 18 Maintaining Audio Quality The full EBU subjective listening test on low bitrate audio codecs study can be downloaded at http tech ebu ch publications tech3296 In 2014 the best overall quality for a given data rate in a transmission codec appears to be achieved by the MPEG AAC codec at rates of 96 kbps or higher and Extended HE AAC at rates below 96 kbps The AAC codec is about 30 more efficient than MPEG1 Layer 3 and about twice as efficient as MPEG1 Layer 2 The AAC codec can achieve transparency that is listeners cannot audibly distinguish the codec s out put from its input in a statistically significant way at a stereo bitrate of 128 kb sec while the Layer 2 codec requires about 256 kb sec for the same quality The Layer 3 codec cannot achieve transparency at any bitrate although its pe
43. NNA and Axia Livewire V2 use this protocol and others has committed to supporting it 23 24 Maintaining Audio Quality AES 67 also includes requirements for interoperability of AVB Layer 2 networks which must be routed using specialized routers and switches There are several other proprietary protocols We expect these to fade away over time because of AES 67 s potential for achieving universal interoperability between various manufacturers products Networked audio connections follow the same exact wiring convention as all other Ethernet data networks Category 5 5e or 6 Ethernet cable should be used Headroom The single most common cause of distorted broadcast netcast sound is probably clip ping intentional in the audio processing chain or unintentional in the program chain In order to achieve the maximum benefit from processing there must be no clipping before the processor The gain and overload point of every electronic com ponent in the station must therefore be critically reviewed to make sure they are not causing clipping distortion or excessive noise In media with limited dynamic range like magnetic tape small amounts of peak clipping introduced to achieve optimal signal to noise ratio are acceptable Never theless there is no excuse for any clipping at all in the purely electronic part of the signal path since good design readily achieves low noise and wide dynamic range Check the following com
44. Radio and WJR Radio at the exact same time He switched between the stations numerous times fascinated by how different the two sounded He could not understand why WJR was allowed to sound so inferior to the BIG full sound of CKLW whose audi ence share was growing daily This seemingly small experience was about to change Greg s life and would ultimately affect the technical sound of radio broadcasting and Internet netcasting as we now know it It took several years of inspiration and hearing the hits from Motown and the LA Wrecking Crew before he understood what would be needed to craft a broadcast audio processing system that would make this music shine over the air by creating a big consistent sound that lived up to or surpassed the standards set by CKLW His first radio gig was at WWWW aka W4 in Detroit while he was still in high school 71 72 Maintaining Audio Quality Here he learned the ins and outs of the commercial broadcast business including the technical workflow In 1975 he founded Gregg Laboratories a broadcast audio signal processing com pany and he has had considerable experience designing commercial broadcast au dio processing systems that many high profile broadcasters have used He has exten sively researched the characteristics of consumer radio receivers and with Robert Orban co authored a technical paper proposing the standardization of pre emphasis in AM broadcast This paper was presented be
45. Robert Orban Greg Ogonowski Maintaining Audio Quality in the Broadcast Netcast Facility 2014 Edition orban Robert Orban Greg Ogonowski Maintaining Audio Quality in the Broadcast Netcast Facility 2014 Edition orban Orban Optimod and Opticodec are registered trademarks All trademarks are property of their respective companies Copyright 1982 2014 Orban orban 8350 E Evans Rd Suite C4 Scottsdale AZ 85284 USA Phone 1 480 403 8300 Fax 1 480 403 8301 E Mail support orban com Web www orban com Table of Contents MAINTAINING AUDIO QUALITY IN THE BROADCAST FACILITY c cccsseceeeeecseceeeeecsaeceeeeecnaeeenees 1 Authors NOte Aevacesdcvecscscazectcesstves ee ic ti A t 1 Introduction araen e E E E EAT 1 PART TRECORDING MEDIA st6sscsiestsnsscibtesessnssgenssatenduetbsceseteisentvde dueratensoeasundhsensedisaasentebterdeenen ies 4 COMPACE DIS Cosita is ire 4 Ouality Controlin CD Transfers avance das ainia e ani idad 7 CD R and CD RW DVD R DVD RW DVD A HD DVD Blur AY coonnnninnnninninncncccneos 9 DIBUAL TAP eC sits ii da O Sills E E E clases a e EE a dessa sty cette 10 Hard DIESE do 11 Flash RAM raised 12 Data Compres SiO orroe oeii enean oE E R E EEE TEE E E AEE E A RER a 12 Lossless COmpresSiOni sss cccs eiscsscscecevecesessnsescestveas ce cebssvses n inise E E E E KEER Ei A 12 Lossy COMPTE E E E E EET 13 MPEG L ayet UA EES 14 AAC HESA AC adoos tha E 15 Memb
46. This time base correction usually occurs in the digital input receiver although further stages can be used downstream Sample rate converters can introduce jitter in the digital domain because they re sample the signal much like A D converters Maintaining lowest jitter in a system requires synchronizing all devices in the audio chain to a common wordclock or AES11 signal This eliminates the need to perform cascaded sample rate conversions on the signals flowing through the facility Good wordclock generators have very low jitter also known as phase noise and allow the cascaded devices to perform at their best There are several pervasive myths regarding digital audio One myth is that long reconstruction filters smear the transient response of digital audio and that there is thus an advantage to using a reconstruction filter with a short impulse response even if this means rolling off frequencies above 10 kHz Several commercial high end D to A converters operate on exactly this mis taken assumption This is one area of digital audio where intuition is particularly deceptive The sole purpose of a reconstruction filter is to fill in the missing pieces between the digital samples These days symmetrical finite impulse response filters are used for in the Broadcast Netcast Facility this task because they have no phase distortion The output of such a filter is a weighted sum of the digital samples symmetrically surrounding
47. agnetic disk flash RAM and data compression are discussed in the body of this document see page 4 while vinyl disk phono graph equipment selection and maintenance analog tape tape recorder main tenance recording alignment tapes and cart machine maintenance are discussed in Appendix Analog Media starting on page 56 System considerations headroom audio metering voice music balance and electronic quality see page 22 The production studio choosing monitor loudspeakers loudspeaker location and room acoustics loudspeaker equalization stereo enhancement other pro duction equipment and production practices see page 43 Equipment following OPTIMOD encoders exciters transmitters and anten nas see page 50 NOTE Because the state of the art in audio technology is constantly advancing it is important to know that this material was last revised in 2014 Our comments and recommendations obviously cannot take into account later developments We have tried to anticipate technological trends when that seemed useful Maintaining Audio Quality Part 1 Recording Media Compact Disc The compact disc CD is currently the primary source of most recorded music for broadcasting With 16 bit resolution and 44 1 kHz sample rate it represents the ref erence standard source quality for radio although it may be superseded in the fu ture by Blu ray which is capable of 24 bit resolution and sample rates up to 192 kHz DVD
48. and decode cycles because each cycle moves the added noise closer to the threshold where the added noise is heard This is particularly critical if the transmission medium itself such as DAR satellite broadcasting or netcasting uses lossy compression Pitfalls When Using Dolby AC3 as a Contribution Codec Although the Dolby AC3 codec was designed as a transmission codec some facilities use it as a contribution codec Dolby recommends Dolby E as a contribution codec and depreciates the use of AC3 for this task In addition to the obvious issue of cascaded codecs AC3 has a tricky potential pitfall when used as a contribution codec AC3 metadata data about the data transmitted as part of the AC3 bitstream in cludes dynamic range control words Dolby s intent was to create dynamics com 21 Maintaining Audio Quality pression that is entirely under the listener s control In essence the AC3 encoder ap plies its input signal to a wideband compressor and transmits the compressor s gain control signal along with the uncompressed audio that was applied to the input of the compressor This way the listeners can enjoy the full dynamic range of the origi nal signal or can apply compression if they prefer a smaller dynamic range To prevent audible wideband gain pumping the amount of dynamic range com pression available from AC3 was purposely limited Many consumers prefer audio dynamic range to be controlled more
49. are female A secret weapon for a very distinc tive sound can be a multiband microphone processor Part 4 Equipment Following OPTIMOD Some of the equipment following OPTIMOD in the transmission path can also affect quality The STL FM exciter transmitter and antenna can all have subtle yet audi ble effects STL The availability of uncompressed digital STLs using RF signal paths has removed one of the major quality bottlenecks in the broadcast chain These STLs use efficient mo dem style modulation techniques to pass digitized signals with bit for bit accuracy If the user uses their digital inputs and outputs and does not require them to do sam ple rate conversion which can introduce overshoot if it a downward conversion that filters out signal energy they are essentially transparent Uncompressed digital STLs using terrestrial lines like T1s in the United States also provide transparent quality and are equally recommended Some older digital STL technology uses lossy compression If the bitrate is sufficiently high these can be quite audibly transparent However all such STLs introduce over shoot and are therefore unsuitable for passing processed audio that has been previ ously peak limited Analog microwave STLs provide far lower quality than either digital technology and are not recommended when high audio quality is desired They are sometimes ap propriate for AM because receiver limitations will tend to mask qual
50. ated that A D should not be used for transfer work Dither is random noise that is added to the signal at approximately the level of the least significant bit It should be added to the analog signal before the A D con verter and to any digital signal before its word length is shortened Its purpose is to linearize the digital system by changing what is in essence crossover distortion into audibly innocuous random noise Without dither any signal falling below the level of the least significant bit will disappear altogether Dither will randomly move this signal through the threshold of the LSB rendering it audible though noisy Whenever any DSP operation is performed on the signal particularly decreasing gain the resulting signal must be re dithered before the word length is truncated back to the length of the input words Ordinarily correct dither is added in the A D stage of any competent commercial product performing the conversion However some products allow the user to turn the dither on or off when truncating the length of a word in the digital domain If the user chooses to omit adding dither this should be because the signal in question already contained enough dither noise to make it unnecessary to add more Many computer software volume controls do not add dither when they attenuate the signal thereby introducing low level truncation distortion It is wise to bypass computer volume controls wherever possible and if this
51. ca tions As of 2014 test tapes are still available from Magnetic Reference Labora tory MRL 165 Wyandotte Dr San Jose CA 95123 www mrltapes com in the Broadcast Netcast Facility 4 Measure flutter Routine maintenance should include measurement of flutter using a flutter me ter and high quality test tape Deterioration in flutter performance is often an early warning of possible mechanical failure Spectrum analysis of the flutter can usually locate the flutter to a single rotating component whose rate of rotation corresponds to the major peak in the filter spectrum Deterioration in flutter per formance can at very least indicate that adjustment of reel tension capstan ten sion reel alignment or other mechanical parameter is required 5 Measure frequency response and interchannel phase shifts These measurements which should be done with a high quality alignment tape can be expedited by the use of special swept frequency or pink noise tapes avail able from some manufacturers like MRL The results provide an early indication of loss of correct head azimuth or of headwear The swept tapes are used with an oscilloscope the pink noise tapes with a third octave real time analyzer The head must be replaced or lapped if it becomes worn Do not try to compen sate by adjusting the playback equalizer This will increase noise unacceptably and will introduce frequency response irregularities because the equalizer can
52. cecssceseesesseesesseesecnseeeceseeseesesseeecueeeecnseesenseeaeeas 67 Sticky Shed Syndrome A 68 Cartridge Tape Machine Maintenance cccccccecceseescesesseesetieeeecuseeecseescnseeseeseeseeseeaeeneeeeenteats 69 ABOUT THE AUTHORS 2053 ccs tees eoa testes A ted eters A A Ea 70 Robert Orbarie nee Meshes A A A dae dc Ay tates AS A Aastats 70 Maintaining Audio Quality in the Broadcast Facility By Robert Orban and Greg Ogonowski Orban Authors Note In 1999 we combined and revised two previous Orban publications on maintaining audio quality in the FM and AM plants and have since revised the resulting publica tion several times In 2014 considerations for both AM and FM are essentially iden tical except at the transmitter because with modern equipment there is seldom rea son to relax studio quality in AM plants The text emphasizes FM HD Radio and net casting practice differences applicable to AM have been edited into the main text Introduction Audio processors change certain characteristics of the original program material in the quest for positive benefits such as increased loudness improved consistency and absolute peak control to prevent distortion in the following signal path and or to comply with government regulations The art of audio processing is based on the idea that such benefits can be achieved while giving the listener the illusion that nothing has been changed Successful au dio processing performs t
53. d follow the manufacturer s instructions Place the analyzer test microphone about 1m from the monitor speaker Adjust the equalizer see its operating manual for instructions to obtain a real time analyzer read out that is flat to 5 kHz and that rolls off at 3dB octave thereafter A truly flat response is not employed in typical loudspeakers and will make most recordings sound unnaturally bright and noisy Electronic equalization cannot fix acoustic nulls in the room caused by standing waves Nulls should be corrected by acoustic treatment of the room and by careful placement of the loudspeakers A good rule of thumb is never to set an equalizer to create a large narrowband boost because this added energy will probably sound unnatural elsewhere in the room where the null does not exist If the two channels of the equalizer must be adjusted differently to obtain the de sired response from the left and right channels suspect room acoustic problems or poorly matched loudspeakers The match is easy to check just physically substitute one loudspeaker for the other and see if the analyzer reads the same Move the microphone over a space of two feet or so while watching the analyzer to see how much the response changes If the change is significant then room acoustic prob lems or very poorly controlled loudspeaker dispersion is likely If it is not possible to correct the acoustic problem or loudspeaker mismatch directly you should at least m
54. d as their software realizations and there are many poor im plementations available especially from the unlicensed open source software com munity DAB formerly called Eureka147 uses the MPEG 1 Layer 2 codec commonly called MP2 This provides poor audio fidelity at 128 kbps and borders on unacceptable at rates of 96 kbps and below Because of these problems DAB has recently been upgraded to DAB which uses the HE AACv2 codec to achieve much more RF spec tral efficiency than DAB by putting three good sounding stereo channels where one mediocre sounding channel used to fit with DAB HD Radio uses a proprietary codec called HDC iBiquity has not released details about it although it is known to use some sort of Spectral Band Replication tech nology see page 19 Its subjective performance is better than MP3 but not as good as HE AACv1 or v2 Audio Processing for Low Bitrate Digital Transmissions It is important to minimize audible peak limiter induced distortion when one is driv ing a low bitrate codec because one does not want to waste precious bits encoding the distortion Look ahead limiting can achieve this goal hard clipping cannot One can model any peak limiter as a multiplier that multiplies its input signal by a gain control signal This is a form of amplitude modulation Amplitude modulation produces sidebands around the carrier signal In a peak limiter each Fourier com ponent of the input signal is a separate
55. d plain AAC bit streams Every decoder is able to handle bit streams of any encoder although a given de coder may not exploit all of the stream s advanced features An HE AACv2 decoder can fully exploit any data inside the bit stream be it plain AAC HE AACv1 AAC SBR or HE AACv2 AAC SBR PS An AAC decoder decodes the AAC portion of the bit stream not the SBR portion As a result the output of the decoder is bandwidth limited as the decoder is not able to reconstruct the high frequency range represented in the SBR data portion of the bit stream If the bitstream is HE AACv2 an AAC decoder will decode it as limited bandwidth mono and an HE AACv1 decoder will emit a full bandwidth mono signal an HE AACv2 decoder is required to decode the parametric stereo information Extended HE AAC xHE AAC is the latest upgrade to the MPEG AAC family It sig nificantly improves the audio quality of music and speech particularly at very low bitrates of 8 kbit s and more and is compatible with HE AAC streams It combines and improves upon HE AACv2 for music and generic audio and AMR WB for speech Extended HE AAC combines the advantages of existing speech and music codecs By adding a new set of encoding tools to the HE AACv2 audio codec Extended HE AAC outperforms dedicated speech and general audio coding schemes and bridges the gap between both worlds providing consistent high quality audio for all signal types Accordingly Extended HE AAC can i
56. dreds of passes shows that it is physically possible for R DAT to be reliable and durable Nevertheless we now advise broadcasters not to trust the reli ability of DAT tape for long term storage and never to use it for new recordings Always make a backup particularly because DAT is now an obsolete format and finding players in working order is more and more difficult If your facility has DAT tapes in storage it would be wise to copy them to other medium as soon as practi cal Hard Disk Systems Hard disk systems use sealed Winchester hard magnetic disks or optical disks origi nally developed for mass storage in data processing to store digitized audio This technology has become increasingly popular as a delivery system for material to be aired There are many manufacturers offering systems combining proprietary soft ware with a bit of proprietary hardware and a great deal of off the shelf hardware If they are correctly administered and maintained these systems are the best way to ensure high consistent source quality in the broadcast facility because once a source is copied onto a hard drive playout is consistent There are no random cueing varia tions and the medium does not suffer from the same casual wear and tear as CDs Of course hard drives fail catastrophically from time to time but RAID arrays can make a system immune to almost any such fault It is beyond the scope of this document to discuss the mechanics of digital deliver
57. e advantages of economy and freedom from mistracking due to level mismatches between record and playback Dolby A was the original multiband compander and many legacy recordings were recorded using the Dolby A system which provides modest amounts of noise reduction but no audible noise pumping or breathing when properly aligned 21 http en wikipedia org wiki Reel to reel audio tape recording 63 64 Maintaining Audio Quality Remember that to achieve accurate Dolby tracking record and playback levels must be matched within 2dB Dolby noise for SR operations or the Dolby tone for Dolby A operations should always be recorded at the head of all reel to reel tapes and level matching should be checked frequently There should be no problem with level matching if tape machines are aligned every week as level standardization is part of this procedure If a different type of tape is put in service recording ma chines must be aligned to the new tape immediately before any recordings are made In our opinion all single ended dynamic noise filter noise reduction systems can cause undesirable audible program dependent side effects and cannot safely be used on line The best DSP based systems can be very effective in the production stu dio where they can be adjusted for each piece of program material but even there they must be used carefully with their operation constantly monitored by the sta tion s golden ears So
58. e attractive as audio sources and for archiving They have error detection and correction built in so when they working correctly their outputs are bit for bit identical to their inputs There are several dye formulations available and manufacturers disagree on their archival life However it has been extrapolated that any competently manufactured CD R should last at least 30 years if it is stored at moderate temperatures below 24 C 75 F and away from very bright light like sunlight On the other hand these 10 Maintaining Audio Quality disks can literally be destroyed in a few hours if they are left in a locked automobile exposed to direct sunlight The industry has less experience with more recent for mats like DVD R and Blu ray No recordable optical medium should be considered to be archival without careful testing Archiving CD R in data format is better than archiving in Red Book audio format because the error correction in data format is more robust Not all media of a given type are equal Choose media to minimize bit error rate BER At the time of this writing Taiyo Yuden TDK and Verbatim are known to have low BER However manufacturers will change formulations and plants from time to time so these recommendations may not be valid in the future The reflectivity of a good CD R is at best 90 of a mass produced aluminized CD Most CD players can accommodate this without difficulty although some very old player
59. e detected even under a microscope because the cause of the degra dation is usually deterioration of the mechanical damping and centering system in the stylus or actual bending of the stylus shank rather than diamond wear This deterioration is primarily caused by back cueing although rough handling will always make a stylus die before its time Styli used in 24 hour service should be changed every two weeks as a matter of course whatever the expense DJs and the engineering staff should listen con stantly for audible deterioration of audio quality and should be particularly sen sitive to distortion caused by a defective stylus Immediately replace a stylus when problems are detected One engineer we know destroys old styli as soon as he replaces them so that he is not tempted to keep a stock of old deteriorated but usable looking styli It is important to maintain a stock of new spare styli for emergencies as well as for routine periodic replacement There is no better example of false economy than waiting until styli fail before ordering new ones or hanging onto worn out 59 Maintaining Audio Quality styli until they literally collapse Note also that smog and smoke laden air may seriously contaminate and damage shank mounting and damping material Some care should be used to seal your stock of new styli to prevent such damage 8 Consider using noise reduction to improve the sound of damaged re cords Several impuls
60. e digital domain and are there fore likely to excite the 0 dBFS phenomenon When using computer sound card analog outputs it is a good idea to make sure the audio levels are no higher than 3dBFS This means that when you rip CDs into a playout system that uses the analog outputs of a sound card you should reduce the level of the audio by 3 dB Before you do this it is important to verify that the DSP implementing the level change adds appropriate dither see page 37 If the bit Maintaining Audio Quality depth is held constant failure to add dither before a gain reduction in the digital domain will introduce distortion If the gain reduction occurs in the analog domain this should not be a problem because any properly designed analog to digital con verter following the analog gain adjustment will add dither as necessary Conventional wisdom holds that pure digital connections from a CD player or a digi tal domain rip cannot cause headroom problems However O dBFS can also be a problem in the digital domain Passing a digitally clipped signal through a sample rate converter even one whose output sample rate is the nominally the same as the input sample rate can cause overshoots because the SRC interpolates samples be tween those existing at the input and the interpolated samples can have a higher level than the input samples Therefore even digital signal paths can cause the 0 dBFS problem so competently designed digital systems
61. e loudspeakers are the single most important influence on studio quality The production studio monitor system is the quality reference for all production work and thus for the final sound to be broadcast netcast Achieving accurate monitor sound requires considerable care in the choice of equipment and in its adjustment Loudspeakers should be chosen to complement room acoustics The space limitations in production studios usually dictate the use of bookshelf sized speakers You should assess the effect of equalization or other sweetening on small speakers to make sure that excessive bass or high frequency boost has not been introduced While such equalization errors can sound spectacular on big wide range speakers it can make small speakers with limited frequency response and power handling capacity sound terrible The Auratone Model 5C Super Sound Cube has frequently been used as a small speaker reference Although these speakers are no longer manufactured they are often available on the used market We recommend that every production stu dio be equipped with a pair of these speakers or something similar and that they be regularly used to assure the production operator that his or her work will sound good on small table and car radios The primary monitor loudspeakers should be chosen for e high power handling capacity e low distortion e high reliability and long term stability e controlled dispersion omnidirectional speakers are not rec
62. e noise reduction systems can effectively reduce the effects of ticks and pops in vinyl disk reproduction without significantly compromising audio quality They are particularly useful in the production studio where they can be optimized for each cut being transferred to other media With the advent of plug in signal processing architectures for both the PC and Mac platforms DSP based signal processing systems have become available at reasonable cost to remove ticks scratches and noise from vinyl disk reproduc tion In a paper like this designed for reasonably long shelf life we can make no specific recommendations because the performance of the individual plug ins is likely to improve quickly These plug ins typically cost a few hundred dollars making them affordable to any production facility Examples of affordable na tive restoration suites include DC8 and iZotope Rx3 8 In addition to impulse noise reduction such suites usually include an FFT based dynamic noise reduction system to reduce low level crackle hiss and rumble These noise reduction systems typically use anywhere from 512 to 2048 fre quency bands enabling them to distinguish between noise and program mate rial in a fine grained manner Most of the systems require the user to provide a noise print of typical noise taken from a part of the groove with no program modulation although the most advanced algorithms also provide a way to automatically estimate
63. e or 20dBFS using a 1kHz sinewave e In installations where the program line has 18 20dB of headroom typical of good engineering practice the Low Level Audio Alarm may often be in the Alarm state e The Digital PPM Low Level Audio Alarm has somewhat inconsistent behav ior It takes as long as 30 to 60 seconds to reset from a Low Signal Level con dition from no audio to normal level audio and takes as long as 10 minutes to enter Low Signal Level condition after a transition occurs between normal audio level and no audio e The Digital PPM encoder continues to encode under Low Signal Level alarm conditions e The alarm is independent of PPM encoder injection functions When audio dynamic range compression not to be confused with bit reduction compression is used after the PPM encoder the low level audio alarm appears to be unimportant because the compression will maintain both modulation levels and PPM signal levels The encoder PPM signal injection is linearly proportional to the audio level at the encoder s input Therefore the location of the PPM encoder before or after the dy namic range compressor does not materially affect data throughput If the com pressor is placed after the PPM encoder as it is in most facilities the audio signal processing systems will amplify the PPM signal and program audio equally so the location will not affect the PPM encoded signal except for a slight inaudible signal to noise penalty H
64. e speaker to the listener and the distance from the speaker to the reflective surface and back to the listener is at least 20 feet 6 meters It is also desirable that the reflections delayed more than 20 milliseconds be well diffused that is with no flutter echoes Flutter echoes are usu ally caused by back and forth reflections between two parallel walls and can often by treated by applying Sonex or other absorbing material to one wall In addition quadratic residue diffusors manufactured by RPG Diffusor Systems Inc can be added to the room to improve diffusion and to break up flutter echoes An excellent short introduction to the theory and practice of LEDE design is Don Davis s article The LEDE Concept in Audio Vol 71 Aug 1987 p 48 58 For a more definitive discussion see Don and Carolyn Davis The LEDE Concept for the Control in the Broadcast Netcast Facility of Acoustic and Psychoacoustic Parameters in Recording Control Rooms J Audio Eng Soc Vol 28 Sept 1980 p 585 95 It should be noted that the LEDE technique is by no means the only way to create a good sounding listening environment although it is perhaps the best documented and has certainly achieved what must be described as a quasi theological mystique amongst some of its proponents Examples of other approaches are found in the August 1987 vol 29 no 8 issue of Studio Sound which focused on studio design Loudspeaker Equalization
65. easure the response at several positions and average the results Microphone mul tiplexers can automatically average the outputs of several microphones in a phase insensitive way they will help you equalize loudspeaker response properly Although left and right equalizers can be adjusted differently below 200Hz they should be set close to identically above 200Hz to preserve stereo imaging even if this results in less than ideal curves as indicated by the third octave analyzer This is a limitation of the third octave analyzer which cannot distinguish between direct sound early reflections and the reverberant field stereo imaging is primarily de termined by the direct sound A few companies are now making DSP based room equalizers that attempt to cor rect both the magnitude and phase of the overall frequency response in the room These can produce excellent results if the room is otherwise acoustically well be haved Recently several companies have developed room correction equalizers that rely on several measurements at different locations in the room They claim that their software can process the results of the multiple measurements to avoid equalizing localized acoustic anomalies 13 For example http www audyssey com in the Broadcast Netcast Facility Finally we note once again that the manufacturers of powered nearfield monitors have done much of the work for you These monitors have built in equalization which
66. ed PreCode This manipulates several aspects of the audio to minimize artifacts caused by low bitrate codecs ensuring consistent loudness and texture from one source to the next Pre Code includes special audio band detection algorithms that are energy and spectrum aware This can improve codec performance on some codecs by reducing audio proc essing induced codec artifacts even with program material that has been preproc essed by other processing than Optimod Summary Maintaining a high level of broadcast netcast audio quality is a very difficult task requiring constant dedication and a continuing cooperation between the program ming engineering and computer IT departments With the constantly increasing quality of home and mobile receivers and stereo gear the broadcast audience more and more easily perceives the results of such dedication and cooperation One suspects that in the future FM DAR and netcasts will have to deliver ever increasing quality to compete successfully with the many other program sources vying for audience attention including CD s DVD s Blu ray disks digital audio subscription television direct satellite broadcast DTV streaming programming on the Internet high resolution downloads and who knows how many others The human ear is astonishingly sensitive perceptive people are often amazed when they detect rather subtle improvements to the broadcast audio chain while listening to an inexpensive car
67. ed by their location in the room Bass is weakest when the speaker is mounted in free air away from any walls bass is most pronounced when the speaker is mounted in a corner Corner mounting should be avoided because it tends to excite standing waves The best location is probably against a wall at least 18 inches 45 cm from any junction of walls If the bass response is weak at this location because the speaker was designed for wall junction mounting it can be corrected by equalization discussed below It is important that the loudspeakers be located to avoid acoustic feedback into the turntable because this can produce a severe loss of definition a muddy sound Many successful monitoring environments have been designed according to the LiveEnd Dead End LEDE concept invented by Don Davis of Synergistic Audio Concepts Very briefly LEDE type environments control the time delay between the arrival of the direct sound at the listener s ear and the arrival of the first reflections from the room or its furnishings The delay is engineered to be about 20 millisec onds This usually requires that the end of the room at which the speakers are mounted be treated with a sound absorbing material like Sonex so that essentially no reflections can occur between the speakers output and the walls they are mounted on or near Listeners must sit far enough from any reflective surface to en sure that the difference between the distance from th
68. ers of the HE AAC Codec Family cecceccsesscesecsseeseeeeceseceeeseesecaecaaeeseeseceaecaeeesenaeceaeeneeneens 15 Independent quality evaluations of AAC HE AACoooocococonononnnonocononnnonncnnocononnnonnonnc nono nono non nc neon nonnnns 17 Spectral Band Replication ceceecccesscesscesscssesseeeecesecaeeeseeseceaecnsecseeseceaecasecseeseeaecaaeaaecaeeeeeeaeeaeeseees 18 Parametric Stereo MPEG Sutroutid ecc de cidiedeseve shan cleeielevanceh at sbtndndestvest ct Gusteleceaceh lala dada Using Data Compression for Contribution cccceccceseesseescesecseesceseceaecaeeeseesecaecaeeesceseeseceaeeneenseeaees Pitfalls When Using Dolby AC3 as a Contribution Codec PART 2 SYSTEM CONSIDERATIONS Wiring and Grounding Analog Interconnection Digital Interconnection GYOUNCING coococccoconcconocononnconccnnnnnncnocnnonns Audio over Ethernet Headro0M ooo Bit Accuracy MM CLOVIS esis cased cosh A Maen ds hee Beas Measuring and Controlling Loudness c cccccecesseesesseeeeesseescnseeseesecieesecseeecuseesensesseeesntenseneeets Jones amp Torick CBS Technology Center Meter cccccecseescesecsseeseeeecesecseeeeeeseceaecaeeaeeeeeeaeeeaeeneees BS 1 770 Loudness Meter iet eine re irori a iG Ai da E A deena bd ir 31 Peak Normalization in Audio Editing Programs ecccscessseescesecsseesceeecesecseeeseeseceaecaecaeeeeeeaeeaeeneens 32 Replay A ea oct Mires a as Sea Mas es ae 32 Speech Mu
69. erter until the step crosses one sample point and therefore the time resolution is limited to one sample The problem with this argument is that there is no such thing as an infinite risetime step function in the digital domain To be properly represented such a function has 39 40 Maintaining Audio Quality to first be applied to an anti aliasing filter This filter turns the step into an expo nential ramp which typically has equal pre and post ringing This ramp can be moved far less than one sample period in time and still cause the sample points to change value In fact assuming no jitter and correct dithering the time resolution of a digital sys tem is the same as an analog system having the same bandwidth and noise floor Ultimately the time resolution is determined by the sampling frequency and by the noise floor of the system As you try to get finer and finer resolution the measure ments will become more and more uncertain due to dither noise Finally you will get to the point where noise obscures the signal and your measurement cannot get any finer However this point is orders of magnitude smaller in time than one sam ple period and is the same as in an analog system A final myth is that upsampling digital audio to a higher sample frequency will increase audio quality or resolution In fact the original recording at the original sample rate contains all of the information obtainable from that recording The onl
70. es to achieve high accuracy Instead of being dependent on the precise switching of voltages or currents to achieve accurate conversion the new designs depend on precise timing which is far easier to achieve in production Oversampling simultaneously increases the theoretical signal to noise ratio and pro duces prior to the reconstruction filter within the CD player a signal that has no significant out of band power near the audio range A simple phase linear analog filter can readily remove this power ensuring the most accurate phase response through the system We recommend that CD players used in broadcast employ tech nology of at least this quality when connected to the broadcast facility via analog connections However the engineer should be aware that these units might emit substantial amounts of supersonic noise so the low pass filtering in the transmission audio processor must be sufficient to reject this to prevent aliasing in digital trans mission processors or STLs The broadcast environment demands ruggedness reliability and quick cueing from audio source equipment The CD player must also be chosen for its ability to track even dirty or scratched CDs with minimum audible artifacts and on its ability to re sist external vibration There are dramatic differences between players in these ar eas We suggest careful comparative tests between players using imperfect CDs to determine which players click mute skip or otherwise mis
71. ew audio stack in Windows Vista 7 Legacy audio applica tions use the emulation mode of this new audio stack Unfortunately Microsoft broke the record sample rate converter in Windows 7 and Server 2008 and fixed it in Windows 8 and higher The play sample rate conversion appears fine Microsoft issued a hotfix for the sample rate conversion problem in Windows 7 and Server 2008 The Windows 7 Server 2008 Sample Rate Converter SRC Hotfix improves the performance of the Windows Core Audio Record Sample Rate Converter by greatly reducing aliasing distortion Not all installations will benefit If you need the hotfix see below download it from http support microsoft com kb 2653312 Install it according Microsoft s instructions Any audio record or encode application that uses the Windows Audio Wave DirectSound API and relies on Windows Core Audio for sample rate conversion will benefit from this hotfix This includes most audio re cord encode applications such as audio editors Sony SoundForge Audac ity and most streaming audio encoders that rely on the Windows sample rate converter Adobe Audition CS5 5 is unaffected because it uses its own sample rate conversion in conjunction with the Microsoft WASAPI API first intro duced in Vista which does not support native sample rate conversion Audio applications that use ASIO or any other third party audio API are also unaffected Windows 7 Server 2008 Audio Devices present themselves to
72. fair amount of coding artifacts Each of these factors severely affects the listening experience and is not perceived as high fidelity SBR Spectral Band Replication is a very useful audio coding enhancement tool It can improve the performance of low bitrate audio and speech codecs by either in creasing the audio bandwidth at a given bitrate or by improving coding efficiency at a given quality level SBR can increase the limited audio bandwidth that a conventional perceptual codec offers at low bitrates so that it equals or exceeds analog FM audio bandwidth 15 kHz SBR can also improve the performance of narrow band speech codecs offering the broadcaster and netcaster speech only channels with 12 kHz audio bandwidth used for example in multilingual broadcasting As most speech codecs are very band limited SBR is important not only for improving speech quality but also for improving speech intelligibility and speech comprehension SBR is mainly a post process although the encoder performs some pre processing to guide the decoding process From a technical point of view SBR is a method for highly efficient coding of high frequencies in audio compression algorithms When used with SBR the underlying coder is only responsible for transmitting the lower part of the spectrum The higher frequencies are generated by the SBR decoder which is mainly a post process fol lowing the conventional waveform decoder Instead of transmitting the spectr
73. fast forward or rewind speed 7 Check playback alignment A Coarsely adjust each recorder s azimuth by peaking the level of the 15 kHz tone on the alignment tape Make sure that you have found the major peak There will be several mi nor peaks many dB down but you will not encounter these unless the head is totally out of adjustment B While playing back the alignment tape adjust the recorder s reproduce equal izers for flat high frequency response and for low frequency response that corresponds to the fringing table supplied with the standard alignment tape Fringing is due to playing a tape that was recorded full track on a half track or quarter track head The fringing effect appears below 500Hz and will ordinarily result in an apparent bass boost of 2 3dB at 100Hz Fine azimuth adjustment cannot be done correctly if the playback equal izers are not set for identical frequency response since non identical fre quency response will also result in non identical phase response C Fine adjust the recorder s azimuth This adjustment is ideally made with a full track mono pink noise tape and a real time analyzer If this instrumentation is available sum the two channels together connect the sum to the real time analyzer and adjust the azimuth for maximum high frequency response If you do not have a full track recorder and real time analyzer you could either observe the mono sum of a swept frequency tape and maximize its
74. fore the Society of Automo tive Engineers and National Association of Broadcasters and was the precursor to the AM NRSC standard He has also authored and co authored many other technical papers on various topics relating to the audio and broadcast industry In 1984 he founded Modulation Index a broadcast engineering consultancy and has done studies on broadcast modulation measurement instrumentation and FM modulators including STL s and exciters As a result of these studies he developed modifications for popular monitors STL S and exciters to improve their dynamic transient accuracy and competitiveness and he presented a technical paper before the National Association of Broadcasters regarding these findings He developed audio signal processing algorithms that were later included in all of the current generation of Optimod audio processors As technical director for KTNQ KLVE Heftel Broadcasting Los Angeles from 1985 to 1991 he relocated studio facilities and constructed a new efficient alternative use AM transmission facility As technical director for KBIG KLAC Los Angeles from 1998 to 2000 he installed a new computer network and digital audio delivery system throughout the business and technical facility KBIG KLAC was one of the first radio stations to stream audio on the Internet with high fidelity sound all from internal encoders and servers After Ogonowski joined Orban in 2000 as VP Product Development he led the team t
75. formers For example never leave a line amplifier or compressor on line in test mode because it seems too much trouble to remove it Small stations often sound dramatically superior to their big time rivals because the small station has a simple audio path while the big budget station has put eve rything but the kitchen sink on line The more equipment the station has or can afford the more restraint and self discipline it needs Keep the audio path simple and clean Every amplifier resistor capacitor transformer switch contact patch bay contact etc is a potential source of audio degradation Corrosion of patch bay con tacts and switches can be especially troublesome and the distortion caused by these problems is by no means subtle In digital signal processing devices the lowest number of bits per word neces sary to achieve professional quality is 24 bits This is because there are a number of common DSP operations like infinite impulse response filtering that substantially increase the digital noise floor and 24 bits allows enough headroom to accommo date this without audibly losing quality This assumes that the designer is sophisti cated enough to use appropriate measures to control noise when particularly diffi cult filters are used If floating point arithmetic is used the lowest acceptable word length for professional quality is 32 bits 24 bit mantissa and 8 bit exponent some times called single precision
76. free from multipath then a clearly audible reduction in high frequency grit or distortion is heard when switching from normal to wide mode D Depending on the Region FM uses either 50us or 75us pre emphasis This se verely limits the power handling capability and headroom at high frequencies and requires very artful transmission processing to achieve a bright sound typi cal of modern CDs Even the best audio processors compromise the quality of the high frequencies by comparison to the quality of flat media like DAB HD and satellite radio These limitations have considerable significance in determining the cost effective ness of current broadcast design practice in the Broadcast Netcast Facility Most older broadcast electronic equipment whether tube or transistor is measura bly and audibly inferior to modern equipment This is primarily due to a design phi losophy that stressed ruggedness and RFI immunity over distortion and noise and to the excessive use of poor transformers Frequency response was purposely rolled off at the extremes of the audio range to make the equipment more resistant to RFI Cascading such equipment tends to increase both distortion and audible frequency response rolloffs to unacceptable levels Modern analog design practice emphasizes the use of high slew rate low noise low cost IC operational amplifiers such as the National LM4562 family the Signetics NE5534 family the National L
77. free from www orban com meter Loudness is subjective it is the intensity of sound as perceived by the ear brain sys tem No simple meter whether peak program meter PPM or VU provides a read ing that correlates well to perceived loudness A meter that purports to measure loudness must agree with a panel of human listeners The Orban Loudness Meter receives a stereo or surround up to 7 1 channels signal from any Windows sound device and measures its loudness and level It can simulta neously display instantaneous true peaks as seen after a D A converter digital sam ple peaks VU PPM CBS Technology Center loudness ITU BS 1770 loudness and EBU R 128 Loudness Range The meter includes peak hold functionality that makes the peak indications of the meters easy to see The software has the ability to analyze audio and the audio parts of video files offline for their BS 1770 3 Integrated Loud ness EBU R 128 LRA highest reconstructed peak level and number of reconstructed peaks above 0 dBfs It will graph the BS 1770 3 Integrated Loudness and peak swings of the CBS Loudness Meter as a function of time and can display a histogram of the BS 1770 3 Integrated Loudness Jones amp Torick CBS Technology Center Meter The CBS meter is a short term loudness meter intended to display the details of moment to moment loudness with dynamics similar to a VU meter It uses the Jones amp Torick algorithm Our DSP implementation of t
78. g so called first order highpass noise shaping and implementing this in Orban products that allow dither to be added to their digital output streams First order highpass noise shaping provides a substantial improve ment in resolution over simple white TPF dither but its total noise power is only 3dB higher than white TPF dither Therefore if it is passed through additional signal processing and or an imperfect D A converter there will be little noise penalty by comparison to more aggressive noise shaping schemes One of the great benefits of the digitization of the signal path in broadcasting is this Once in digital form the signal is far less subject to subtle degradation than it would be if it were in analog form although in fixed point form it is still subject to clipping for reasons discussed earlier in this book Short of being clipped or becom ing entirely un decodable the worst that can happen to the signal is deterioration of noise shaped dither and or added jitter Jitter is a time base error The only jitter than cannot be removed from the signal is jitter that was added in the original analog to digital conversion process All subse quent jitter can be completely removed in a sort of time base correction opera tion accurately recovering the original signal The only limitation is the performance of the time base correction circuitry which requires sophisticated design to re duce added jitter below audibility
79. gh loudness material This will cause the meter to under read program material like dialog having substantial pauses that contain only low level ambience because louder program material contributes most to a lis tener s perception of overall program loudness To address this problem the BS 1770 3 algorithm adds gating to the BS 1770 1 algo rithm There are two steps in the gating process first an absolute gate removes si lent passages second a relative gate weights louder parts of the program more heavily that quieter parts A more detailed explanation of the algorithm is this 1 Using the BS 1770 1 algorithm i e a K weighting filter followed by RMS summa tion and averaging calculate the RMS value in a 400 ms time window One number is computed for every 400 ms time window Start computing a new 400ms window every 100 ms so there is 75 time overlap between windows Continue computing the RMS values of new 400ms windows throughout the entire duration of the meas urement and store all of these results one number for each 400ms window 2 If any 400ms window has a value below 70 LKFS throw it away 3 Compute the average of the remaining windows over the total time period of the measurement If any window is less than 10 dB below this average throw it away 4 Compute the average of the remaining windows Display this reading on the me ter 31 32 Maintaining Audio Quality Orban has written a white pape
80. hat created Optimod PC a PCI sound card with onboard audio processing for any digital audio or streaming application He currently oversees the engineering de partment where audio encoders editors and signal processors currently under de velopment will enable Orban to continue in its tradition of high quality high per formance broadcast technology Determined to change the way Internet streaming audio is perceived and consumed he was the architect of the first commercial high quality file and streaming audio encoder using standards based MPEG 4 AAC HE AAC and MPEG Surround Orban Opticodec PC He also created a high quality HE AAC streaming player supporting standards based protocols the StreamS HiFi Radio App which was the first Adobe Flash streaming audio player for Apple iPhone iPad
81. he core codec is stereo allowing MPEG Surround to be completely stereo compatible but a mono core codec can be used at very low bitrates at the expense of reduced subjective performance Using Data Compression for Contribution Using lossy compression to store program material for playout is one area where AM practice might diverge from FM and DAB practice Because of the lower audio reso lution of AM at the typical receiver an AM station trying to economize on storage might want to use a lower data rate than an FM or DAR station However this is likely to be false economy if the owner of this library ever wants to use it for higher fidelity services like netcasting FM or DAR in the future In general increasing the quality reduces the likelihood that the library will cause problems in future Any library recorded for general purpose applications should use at least 44 1 kHz sample rate so that it is compatible with digital radio systems having 20 kHz band width If the library will only be used on FM and AM 32 kHz is adequate and will save considerable storage However given the rise of digital radio and netcasting we cannot recommend that any forward looking station use 32 kHz for storage Be cause CD audio is 44 1 kHz sample rate conversion is unnecessary This eliminates a process that can potentially degrade the audio quality At this writing the cost of hard disks and other digital storage media is declining so rapidly that there
82. he desired electrical modifications while presenting a result to the listener that subjectively sounds natural and realistic This sounds impossible but it is not Audio processing provides a few benefits that are often unappreciated by the radio or television listener For example the reduction of dynamic range caused by proc essing makes listening in noisy environments particularly the car much less difficult In music having a wide dynamic range soft passages are often lost completely in the presence of background noise Few listeners listen in a perfectly quiet environment If the volume is turned up subsequent louder passages can be uncomfortably loud In the automobile dynamic range cannot exceed 20 dB without causing these prob lems Competent audio processing can reduce the dynamic range of the program without introducing objectionable side effects Further broadcast program material typically comes from a rapidly changing variety of sources most of which were produced with no regard for the spectral balances of others Multiband limiting when used properly can automatically make the segues between sources much more consistent Multiband limiting and consistency are vital to the station that wants to develop a characteristic audio signature and strong positive personality just as feature films are produced to maintain a consistent look Ultimately it is all about the listener experience Good broadcast operators are hard to fi
83. his algorithm typically matches the original meter within 0 5 dB on sinewaves tone bursts and noise The original me ter uses analog circuitry and an LED bar graph display with 0 5 dB resolution Many researchers have been curious about the Jones amp Torick meter but been unable to evaluate it and compare it with other loudness meters Orban developed this soft ware because we believed it would be useful to practicing sound engineers and re searchers and because we are using the CBS meter in our Optimod 8585 and 8685 surround audio processors Optimod PC 1101 1101e v2 and Optimod DAB 6300 v2 The Jones amp Torick algorithm improves upon the original loudness measurement algorithm developed by CBS researchers in the late 1960s Its foundation is psycho acoustic studies done at CBS Laboratories over a two year period by Torick and the late Benjamin Bauer After surveying existing equal loudness contour curves and finding them inapplicable to measuring the loudness of broadcasts Torick and Bauer organized listening tests that resulted in a new set of equal loudness curves based on octave wide noise reproduced by calibrated loudspeakers in a semirever berant 16 x 14 x 8 room which is representative of a room in which broadcasts are normally heard They published this work in Researches in Loudness Measure 6 Refer to the USER MANUAL HERE link at www orban com meter for up to date documentation for the meter 7 Jones Bronwyn L To
84. homemade alignment tape will be recorded see step 7 on page 66 While aligning the master recorder write down the actual VU meter reading produced at each frequency on the spot frequency standard alignment tape B Subtract the compensation specified on the fringing table from the VU meter readings taken in step A Because you are recording in half track stereo instead of full track mono you will use these compensated readings when you record your secon dary standard tape 68 Maintaining Audio Quality C Excite the record amplifier of the master recorder with pink noise spot fre quencies or swept tones D Adjust the azimuth of the master recorder s record head by observing the mono sum from the playback head Pink noise and a real time analyzer are most effective for this If the traditional Lissajous pattern is used use several frequencies and adjust for minimum differential phase at all frequencies E Set the master recorder s VU meter to monitor playback F Record your secondary standard alignment tape on the aligned master re corder Use an audio oscillator to generate the spot frequencies Immediately af ter each frequency is switched in adjust the master tape recorder s re cord gain control until the VU meter reading matches the compensated meter readings calculated in step B Your homemade tape should have an error of only 0 5dB or so if you have followed these instructions carefully
85. ic Because lossless audio codecs are transparent their usability can be assessed by speed of compression and decompression compression efficiency robustness error correction file tagging features and software and hardware compatibility Unless there is an error or bug in the implementation of the codec it is almost impossible in the Broadcast Netcast Facility for different lossless codecs to sound different Although one could conceive of a scenario where the different algorithms load a decoding computer s CPU differently and hence introduce different amounts of jitter into an onboard DAC via ground currents or power supply modulation we are unaware of any evidence that this has ever actually been demonstrated Unless an audible difference between lossless compression algorithms survives a double blind listening test it is safe to assume that any such claims have no physical reality and are caused by the expectation or placebo effect in the mind of the listener Lossy Compression mu Lossy compression eliminates data that its designer has determined to be irrele vant to human perception permitting the noise floor to rise instead in a very fre quency dependent way This exploits the phenomenon of psychoacoustic masking which means that quiet sounds coexisting with louder sounds will sometimes be drowned out by the louder sounds so that the quieter sounds are not heard at all The closer in frequency a quiet sound
86. ig time quality to the announcer But be careful not to use too much bass boost because it can de grade intelligibility Effects like telephone and small transistor radio can be achieved with equalization too The punch of production material can often be enhanced by tasteful application of compression to the microphone chain However avoid using an excessive amount of gain reduction and excessively fast release time These cause room noise and announcer breath sounds to be exaggerated to grotesque levels al though this problem can be minimized if the compressor has a built in expander or noise gate function When adjusting the microphone processor adjust the main audio processor for your desired sound on music first and then adjust the microphone processor to complement the main processing you have selected Close micing which is customary in the production studio can exaggerate voice sibilance In addition many women s voices are sibilant enough to cause un pleasant effects High frequency equalization and or compression will further exaggerate sibilance If you prefer an uncompressed sound for production work 49 Maintaining Audio Quality but still have a sibilance problem then consider locating a dedicated de esser af ter all other processing in the microphone chain It might be necessary to use personalized microphone processing settings for different announcers particu larly if some are male and some
87. ile is passed through other digital processing it is subject to the O dBFS issue unless said processing has been designed to allow sufficient headroom The CD CHECK Test Disc is useful to validate a CD player or CD drive error correction and is available from Digital Recording at http www digital recordings com cdcheck cdcheck html It is also possible to extract or rip DVD A audio if multichannel audio is required A very capable application is DVD Audio Extractor and available here http www dvdae com As of this writing SACD DSD cannot be digitally extracted or ripped SACD Hybrid discs with a CD layer can be ripped using the standard CD digital audio extraction method described earlier It is only a matter of time before SACD DSD extraction becomes available In the meantime an analog transfer of the SACD layer is likely to provide the highest audio quality available because the SACD layer is usually al in the Broadcast Netcast Facility though not always mastered without the quality degrading peak limiting and clip ping that have become ubiquitous on contemporary CD releases Quality Control in CD Transfers When one builds a music library on a digital delivery system it is important to vali date all audio sources A track s being available on CD does not guarantee good au dio quality For the best audio quality results do not accept MP3 audio from record companies or download services MP4 AAC audio may be u
88. in double precision floating point arithmetic The resulting error signal was a mini mum of 125 dB below full scale on a sample by sample basis which was comparable to the stopband depth in the experimental reconstruction filter We therefore have the paradoxical result that in a properly designed digital audio system the frequency response of the system and its sound is determined by the anti aliasing filter and not by the reconstruction filter Provided that they are real ized with high precision arithmetic longer reconstruction filters are always better This means that a rigorous way to test the assumption that high sample rates sound better than low sample rates is to set up a high sample rate system Then without changing any other variable introduce a filter in the digital domain with the same frequency response as the high quality anti aliasing filter that would be required for the lower sample rate If you cannot detect the presence of this filter in a double blind test then you have just proved that the higher sample rate has no intrinsic audible advantage because you can always make the reconstruction filter audibly transparent Another myth is that digital audio cannot resolve time differences smaller than one sample period and therefore damages the stereo image People who believe this like to imagine an analog step moving in time between two sample points They argue that there will be no change in the output of the A D conv
89. included Agfa PEM468 Ampex 406 Ampex 456 BASF SPR 50 LHL EMI 861 Fuji type FB Maxell UD XL TDK GX Scotch 3M 206 Scotch 250 Scotch 226 and Sony SLH1 1 In 2014 none of these tapes are in the Broadcast Netcast Facility being manufactured Before considering use of old stock of these tapes for new re cordings be aware that several suffered from sticky shed syndrome see page 68 and may have deteriorated severely since they were manufactured It is safer to use newly manufactured tape for new recordings In 2006 Recorded Media Group International RMGI in the Netherlands began manufacturing EMTEC specification tape in Oosterhout ATR Magnetics of York PA longtime service and modification shop for multitrack and master recorders began manufacturing analog multitrack tape and in November 2006 began beta testing a new formula Jai Electronic Industries in India is currently making audio tape in 6 35 mm 1 4 and 12 7 mm 1 2 width and perforated 16 mm and 35 mm audio tape for the film industry Daniel Technology in the USA are making 3 81 mm tape for the Nagra SN series tape recorders Pyral in France are making perforated 16 mm 17 5 mm and 35 mm audio tape Tape Speed If all aspects of the disk to tape transfer receive proper care then the difference in quality between 15ips 38cm sec and 7 5ips 19cm sec recording is easily audible 15ips has far superior high frequency headroom The effects of drop out
90. io qual ity than any other vinyl playback device known to us Despite its close to the master tape sound quality the laser turntable has several drawbacks It is very sensitive to dust and imperfections in the grooves of a disk so a wet vacuum cleaning using a machine like a Loricraft Nitty Gritty or VPI prior to playback is unconditionally required Of course any archival transfer of vinyl should start with such a cleaning regardless of the playback technology employed The la ser turntable will not play certain out of standard records such as records where the cut starts on the outside raised bead and its trackability is average it will not track extremely high groove velocities that a state of the art cartridge can readily handle Finally it will not track non black vinyl such as picture disks For these rea sons it cannot entirely supplant mechanical playback However it will correctly play a great majority of disks and it can work wonders by ignoring surface damage such as shallow scratches that conventional playback will reveal Another important accessory for the specialist vinyl archiver particularly when using the Laser Turntable is a digital de clicker and noise reduction system See step 8 on page 60 The following should be carefully considered when choosing and installing conven tional vinyl disk playback equipment 1 Align the cartridge with great care When viewed from the front the stylus must
91. ion techniques The cross correlation circuit should be first fol lowed by the pilot tone correction circuit With such an approach any mistakes made by the cross correlation technique would be corrected by the pilot tone tech nique older material without pilot tone encoding would usually be adequately cor rected by cross correlation Encoding all synthesized stereo material with pilot tones would prevent embarrassing broadcast errors Cheap Tape Cheap tape whether reel or cart is a temptation to be avoided Cheap tape may suffer from any or all of the following problems e Sloppy slitting causing the tape to weave across the heads or if too wide to slowly cut away your tape guides e Poor signal to noise ratio e Poor high frequency response and or high frequency headroom e Inconsistency in sensitivity bias requirements or record equalization re quirements from reel to reel or even within a reel e Splices within a reel e Oxide shedding causing severe tape machine cleaning and maintenance problems e Squealing due to inadequate lubrication High end name brand tape is a good investment It provides high initial quality and guarantees that recordings will be resistant to wear and deterioration as they are played Whatever your choice of tape you should standardize on a single brand and type to assure consistency and to minimize tape machine alignment problems Some of the most highly regarded tapes in 1990 use
92. is superficial and futile Processors must be judged on how they perform with the many different types of program material used in a given format and ulti mately should be judged based on their ability to attract and hold a given broad caster s target audience There is no substitute for long term listening AM MW is limited by poor signal to noise ratio and by limited receiver audio band width typically 2 3 kHz As delivered to the consumer it can never be truly high fidelity Consequently multiband audio processing for AM compresses dynamic range more severely than in typical FM or digital practice In addition pre emphasis whether NRSC or more extreme than NRSC is required to ensure reasonably crisp intelligible sound from typical AM MW radios In AM this is always provided in the audio processor and never in the transmitter Audio quality in TV viewing is usually limited by small speakers in the receivers al though the increasing popularity of DTV HDTV personal entertainment and home theatre is changing this increasing consumer demand for high audio quality In eve ryday television viewing it is important to avoid listener irritation by maintaining consistent subjective loudness from source to source A CBS Loudness Controller combined with multi band processing both included in OPTIMOD TV can achieve this Netcasting also known as webcasting DAB and HD Radio almost always use low bit rate codecs Processing for s
93. is to a loud sound the more efficiently the louder sound can mask it There are also temporal masking laws having to do with the time relationship between the quieter and louder sounds A good psychoacoustic model that predicts whether an existing sound will be masked is complicated The interested reader is referred to the various papers on perceptual coders that have appeared since the late 1980s in EBU references and in the professional literature mostly in the Journal of the Audio Engineering Society and in various AES Convention Preprints There are two general classes of lossy compression systems non psychoacoustic and psychoacoustic The first is exemplified by ADPCM and APT X which while de signed with full awareness of psychoacoustic laws do not contain psychoacoustic models In exchange for this relative simplicity they have a very short delay time less than 4ms which is beneficial for applications requiring foldback monitoring for example The second class contains built in psychoacoustic models which the encoder uses to determine what parts of the signal will be thrown away and how much the noise floor can be allowed to rise without its becoming audible More advanced codecs like MPEG 2 AAC contain adaptive filterbanks that minimize audible pre echo on transients These codecs can achieve higher subjective quality for a given bitrate than codecs of the first class at the expense of much larger time delays Examples i
94. iters like the WAVES L1 enabled CD mastering engineers to limit peaks without obvious side effects However like most anything else the availability of the tools led to their abuse Digital limiting which was a bit like crack in the Broadcast Netcast Facility to some mastering engineers started to suck the life and punch out of material for the sake of loudness Certain engineers developed a reputation as go to guys if you wanted a loud CD Look ahead limiting by itself was no longer enough some engineers started to run material through analog clippers prior to A D conver sion This allowed them to chop off peaks caused by snare drums and the like with out the pumping that digital limiters could add to such material Even that wasn t enough Because the life was sucked out of the material by too much clipping and limiting some engineers started tarting up the corpse with yet more signal processing before the peak limiters For examples vacuum tube equal izers and compressors were used to add some sparkle by driving the tubes into soft distortion While this could actually help some material that was too sterile and that didn t sound like a record once again it could be and was abused All too often today s CDs are squashed into fatiguing mush by over processing Fin gernails on a chalkboard brightness combined with dynamic flatness to create sound that many music consumers find disturbing without really knowing
95. ity limitations in the STL Recently the industry has informally implemented a digitized FM composite base band connection using the left channel of a 192 kHz AES3 link Because traditional analog composite connections are simple and robust the main advantages of digi tizing the composite are 1 increasing the resistance of the link to noise and EMI and 2 allowing the entire signal path from studio to transmitter to remain in the digital domain Orban has extended this implementation to 384 kHz sampling by using the right channel of the link to pass even samples of 384 kHz while the left channel passes odd samples This allows the full FM baseband 0 99 kHz to be accommodated on the link If the bandwidth of the original baseband signal is limited to 96 kHz this signal is 100 backward compatible with the implementation that uses the left channel only at 192 kHz in the Broadcast Netcast Facility FM Exciter Exciter technology has improved greatly since FM s early years The most important improvement has been the introduction of digitally synthesized exciters from several manufacturers This technology uses no AFC loop and can have frequency response to DC if desired It therefore has no problems with bounce or tilt to cause overshoot In conventional analog exciter technology the major improvements have been low ered non linear distortion in the modulated oscillator and higher performance Automatic Frequency Control AFC loops
96. llboard Magazine June 5 1999 This article is not available on line unless you subscribe to Billboard s online service so a local library may be the best way of getting it in the Broadcast Netcast Facility 69 will minimize the probability that the tape will suffer catastrophic damage later on Cartridge Tape Machine Maintenance The above comments on tape recorder maintenance apply to cart machines as well However cart machines have further requirements for proper care largely because much of the tape guidance system is located within the cartridge and so is quite sensitive to variations in the construction of the individual carts While few cart machines are still in use some broadcasters have found that the heft ier ones make good doorstops 1 Clean pressure rollers and guides frequently Because lubricated tape leaves lubricant on the pressure rollers and tape guides frequent cleaning is important in achieving the lowest wow and flutter and in preventing possible cartridge jams Cleaning should be performed as often as experience proves necessary Because of the nature of tape lubricant it does not tend to deposit on head gaps so head cleaning is rarely required 2 Check head alignment frequently Even with the best maintenance interchannel phase shifts in conventional cart machines will usually prove troublesome In addition different brands of car tridges will show significant differences in phase stabili
97. me possible applications include noise reduction of outside production work and when placed after the microphone preamp reduction of am bient noise in the control room or production studio Tape Recorder Maintenance Regular maintenance of magnetic tape recorders is crucial to achieving consistently high quality sound Tape machine maintenance requires expertise and experience The following points provide a basic guide to maintaining a tape recorder s per formance 1 Clean heads and guides every four hours of operation 2 Demagnetize heads as necessary Tradition has it that machines should be demagnetized every eight hours In our experience magnetization is usually not a problem in playback only machines in fixed locations A magnetometer with a 5 gauss scale available from R B Annis Co Indianapolis Indiana USA should be used to periodically check for perma nent magnetization of heads and guides You will find out how long it takes for your machines in your environment to pick up enough permanent magnetization to be harmful You may well find that this never happens with playback ma chines Recording machines should be watched much more carefully 3 Measure tape machine performance frequently Because tape machine performance usually deteriorates gradually measure the performance of broadcast machines frequently with standard test tapes Take whatever corrective action is necessary if the machine is not meeting specifi
98. mples in a bitstream without cor rectly predicting the peak level of the reconstructed analog waveform after D A conversion or the peak level of digital samples whose sample rate has been con verted The meter may under read the true peak level by as much as 3 dB This phe nomenon is known as OdBFS The ITU BS 1770 Recommendation Algorithms to measure audio programme loudness and true peak audio level suggests oversam pling a true peak reading meter by at least 4x and preferably 8x By filling in the space between the samples oversampling allows the meter to indicate true peaks more accurately This allows the 0 dBFS phenomenon to be monitored and pre vented For older equipment with very soft clipping characteristics it may be impossible to see a well defined clipping point on a scope Or worse audible distortion may occur many dB below the apparent clip point In such a case the best thing to do is to de termine the peak level that produces 1 THD and to arbitrarily call that level the clipping level Calibrate the scope to this 1 THD point and then make headroom measurements Engineers should also be aware that certain system components like microphone preamps phono preamps and computer soundcards have absolute input overload points Difficulties often arise when gain controls are placed after early active stages because the input stages can be overloaded without clipping the output Many broadcast microphone pream
99. mprove the quality of existing low bitrate services or more audio channels can be transmitted at a given bitrate Standardization AAC HE AAC is an open standard and not a proprietary format unlike other less effi cient codecs It is widely standardized by many international standardization bodies as follows e MPEG 2 AAC 4 Neuendorf Max Multrus Markus Rettelbach Nikolaus Fuchs Guillaume Robil liard Julien Lecomte J r mie Wilde Stephan Bayer Stefan Disch Sascha Helm rich Christian Lefebvre Roch Gournay Philippe Bessette Bruno Lapierre Jimmy Kj rling Kristofer Purnhagen Heiko Villemoes Lars Oomen Werner Schuijers Erik Kikuiri Kei Chinen Toru Norimatsu Takeshi Chong Kok Seng Oh Eunmi Kim Miyoung Quackenbush Schuyler Grill Bernhard The ISO MPEG Unified Speech and Audio Coding Standard Consistent High Quality for All Content Types and at All Bitrates J AES Volume 61 Issue 12 pp 956 977 December 2013 in the Broadcast Netcast Facility e MPEG ISO IEC 13818 7 2004 Advanced Audio Coding e MPEG 4 AAC e MPEG ISO IEC 14496 3 2001 Coding of Audio Visual Objects Audio includ ing Amd 1 2003 Bandwidth Extension Amd 2 2004 Parametric Coding for High Quality Audio and all corrigenda e MPEG 4 HE AACv1 AAC LC SBR aka HE AAC or AAC e MPEG ISO IEC 14496 3 2001 AMD 1 Bandwidth Extension e MPEG 4 HE AACv2 AAC LC SBR PS aka Enhanced HE AAC or eAAC e MPEG ISO I
100. ms regarding loudness and quality Every radio is equipped with a volume control and every listener knows how to use it If the listener has access to the volume control he or she will adjust it to his or her preferred loudness After said listener does this the only thing left distinguishing the sound of the radio station is its texture which will be either clean or de graded depending on the source quality and the audio processing Any Program Director who boasts of his station s 20 000 worth of enhancement equipment should be first taken to a physician who can clean the wax from his ears then forced to swear that he is not under the influence of any suspicious substances and finally placed gently but firmly in front of a high quality monitor system for a in the Broadcast Netcast Facility demonstration of the degradation that 20 000 worth of enhancement causes Always remember that less is more The Arbitron PPM Portable People Meter The Arbitron PPM Encoder is an audio watermarking device that adds encoded data about the program audio to the audio itself so that a monitoring device equipped with a microphone and worn by a listener can receive the data via acoustic trans mission from the radio receiver or computer loudspeaker The PPM algorithm which is proprietary to Arbitron is based on the well known principle of psycho acoustic masking For most listeners the program material masks or drowns out
101. must have enough head room to prevent clipping in the digital domain caused by O dBFS Using a stand alone CD player to source audio for a digital playout system is cur rently a common ways to transfer CD audio to these systems The analog outputs are subject to 0 dBFS as outlined earlier The primary advantages to real time transfers are that the transfer engineer can detect any audio glitches caused by damaged or defective CDs and that is that it is easy to control levels during the transfer This in cludes increasing the level of song intros which is a common practice To achieve the best accuracy use a digital interface between the CD player and the digital playout system An alternative is to extract the digital audio from the CD us ing a computer and a program to rip the audio tracks to a digital file The primary advantage of computer ripping is speed However it is crucial to use the right hard ware and software to achieve error correction equivalent to that routinely found in a stand alone CD player A combination of an accurate extraction program such as Exact Audio Copy or EAC http www exactaudiocopy de and a Plextor CD DVDdrive which implements hardware error correction will yield exceptional results and will automatically log and detect uncorrectable errors Not all drives are capable of digital audio extraction and not all drives offer hardware error correc tion Moreover bear in mind that once the ripped audio f
102. n listens to the sum of the channels and minimizes audible high frequency loss Several manufacturers have sold electronic phase correction devices that they claim eliminate the effects of interchannel phase shifts although to our knowledge none of these is currently being manufactured 20 http www tcelectronic com 61 62 Maintaining Audio Quality One type of phase correction device measures the cross correlation between the left and right channels and then introduces interchannel delay to maximize the long term correlation This approach is effective for intensity stereo and pan potted multitrack recordings that is for almost all pop music but makes frequent mis takes on recordings made with spaced array microphone techniques due to the normal phase shifts introduced by wide microphone spacing and makes disastrous mistakes with material that has been processed by a stereo synthesizer Another type of phase correction device introduces a high frequency pilot tone am plitude modulated at a low frequency into both the left and light channels Al though the accuracy of this approach is not affected by the nature of the program material it does require pre processing of the material adding the pilot tone and so may not be practical for stations with extensive libraries of existing non encoded material It is theoretically possible to use a combination of the cross correlation and pilot tone phase correct
103. nclude the MPEG family of encoders including Layer 2 Layer 3 AAC and HE AAC also known as aacPlus The Dolby AC 2 and AC 3 codecs also fall in this category The large time delays of these codecs make them unsuitable for any application where they are processing live microphone signals that are then fed back into the announcer s headphones In these applications it is sometimes possible to design the system to bypass the codec feeding the undelayed or less delayed signal into the headphones There are two general applications for codecs in broadcasting contribution and transmission A contribution class codec is used in production Accordingly it must have high enough mask to noise ratio that is the headroom between the actual codec induced noise level and the just audible noise level to allow its output to be 13 14 Maintaining Audio Quality processed and or to be cascaded with other codecs without causing the codec induced noise to become unmasked and without introducing audible pre echo A transmission class codec is the final codec used before the listener s receiver Its main design goal is maximum bandwidth efficiency Some codecs like Layer 2 have been used for both applications at different bitrates There are many proprietary non MPEG codecs other than Dolby AC3 available but these are not standards based and are beyond the scope of this document Ideally all codecs implementing a gi
104. nd making artful automatic gain control essential for the correction of errors caused by distractions or lack of skill Also the regulatory authorities in most countries have little tolerance for excessive modula tion making peak limiting mandatory for signals destined for the regulated public airwaves Maintaining Audio Quality OPTIMOD FM OPTIMOD AM OPTIMOD DAB OPTIMOD TV and OPTIMOD PC have been designed to meet the special problems and needs of broadcasters and net casters while delivering a quality product that most listeners consider highly pleas ing However every electronic communication medium has technical limits that must be fully heeded if the most pleasing results are to be presented to the audience For instance the audio quality delivered by OPTIMOD is highly influenced by the quality of the audio presented to it If the input audio is very clean the signal after process ing will probably sound excellent even after heavy processing Distortion of any kind in the input signal is likely to be exaggerated by processing and if severe can end up sounding offensive and unlistenable Audio processing is an art and the sound of a given audio processor is a function of hundreds of variables many of which involve trade secrets known only to their manufacturers This includes Orban Comparing audio processors by counting the number of bands of compression limiting or listing other features obvious from the front panel
105. o occur in the transmission chain after the analog encoder Correcting this requires the gain after the encoder to be lowered which reduces the loudness of the transmission and partially defeats the purpose of the audio processing system The PPM encoder uses an elegant psychoacoustic audio masking model to combine digital data called a watermark with the actual audio signal The same principle is used in perceptual audio codecs By capturing measuring processing and analyz ing signals from a Digital Arbitron PPM encoded broadcast program line without attempting to reverse engineer the bitstream data format we were able to deter mine the following details e There are 10 frequency bins which are located between 1 and 3kHz e PPM decoders are very pitch sensitive This is very important to consider when using PPM encoded signals for netcasting using Adobe Flash Players Numerical inaccuracies in the sample rate converter of the Adobe Flash en coder and player render the PPM signal useless at sample rates other than 41 42 Maintaining Audio Quality 44 1kHz This is unfortunate because a 32kHz sample rate optimizes the per formance of HE AACv2 low bitrate netcasting e The encoded output signal is the ratio of the input audio signal and the PPM signal e The Digital PPM encoder has a five sample delay at 44 1kHz 113 38us e The Digital PPM Low Level Audio Alarm threshold is 18dBFS using normal ized pink nois
106. olume control in series with the volume control available to the consumer Use of Dialnorm thus allows the loudness of programs from various providers and sources to be consistent regardless of their choice of reference level The ATSC Recommended Practice a_85 2009 Techniques for Establishing and Maintaining Audio Loudness for Digital Television offers an extensive discussion of production and transmission techniques that can be used to maintain consistent loudness at a consumer s digital television receiver regardless of program source Many systems use digital audio sound cards to get audio signals in and out of com puters that are used to store process and play audio However not all sound cards have equal performance even when using digital input and output For example a sound card may unexpectedly change the level applied to it Not only can this de stroy system level calibration but gain can introduce clipping and loss can introduce truncation distortion unless the gain scaled signal is correctly dithered If the analog input is used gain can also introduce clipping and in this case loss can compromise the signal to noise ratio Further the A D conversion can introduce nonlinear distor tion and frequency response errors In almost all modern professional facilities reference level 4 dBu and circuits clip at 20 dBu or higher When using analog I O consumer and prosumer computer 5 www atsc org cms standards a_
107. ommended e good tone burst response at all frequencies e lack of cabinet diffraction e relatively flat axial and omnidirectional frequency response from 40 15 000Hz e physical alignment of drivers when all drivers are excited simultaneously the 44 Maintaining Audio Quality resulting waveforms should arrive at the listener s ears simultaneously some times called time alignment There are a number of powered midfield monitors available from a large assortment of pro audio companies like JBL KRK Focal Mackie Genelec Tannoy and Alesis among others These speakers are very convenient to use because they have built in power amplifiers and equalizers Because they have been designed as a system they are more likely to be accurate than random combinations of power amplifiers equalizers and passive loudspeakers They are also less likely to be connected out of phase which will cause dramatic loss of low frequencies vague stereo imaging and an overall hollow sound The principal influence on the accuracy of these powered speakers particularly at low frequencies is room acoustics and where the speakers are placed in the room Some of these speakers allow the user to set the bass equali zation to match the speaker s location We believe that such speakers are a logical choice for main monitors in a broadcast production studio Loudspeaker Location and Room Acoustics The bass response of the speakers is strongly affect
108. ow as many users as possible in that network cell to access mo bile communication services in parallel Highly efficient speech and audio codecs al low operators to use their spectrum most efficiently Considering the impact that the advent of multimedia services has on the data rate demands in mobile commu nication systems it is clear that even with 4G LTE 3GPP CDMA2000 EDGE and UMTS cellular networks will find it necessary to use perceptual codecs at a relatively low data rate Although many wireless carriers claim to provide high data rates multimedia requires a consistent data rate to prevent media dropouts Low bitrate codecs prevent dropouts on congested networks Using perceptual codecs at low bitrates however has a downside State of the art perceptual audio codecs such as AAC achieve CD quality or transparent audio quality at a bitrate of approximately 128 kbps 12 1 compression Below 96 kbps the perceived audio quality of most of these codecs begins to degrade significantly Either the codecs start to reduce the audio bandwidth and to modify the stereo im age or they introduce annoying coding artifacts caused by a shortage of bits when they attempt to represent the complete audio bandwidth Both ways of modifying in the Broadcast Netcast Facility the perceived sound can be considered unacceptable above a certain level At 64 kbps for instance AAC either would offer an audio bandwidth of only 12 5 kHz or introduce a
109. owever placing the compressor before the PPM encoder will pre vent the low level audio alarm from tripping because the compression keeps the level constant at the input of the PPM encoder While the PPM system attempts to increase ratings accuracy by replacing the old pa per diary reporting system with an automated one there are a number of technical reasons why this is not completely foolproof High average audio levels are required to maintain PPM signal integrity Program material with many quiet periods such as talk formats offers less opportunity to produce an encoded signal This is a limitation of this kind of technology It is clear that using considerable amounts of audio dynamic range reduction can help maximize PPM data throughput and give stations higher ratings Orban Opti in the Broadcast Netcast Facility mods can provide high average audio levels for PPM encoders without introducing side effects that drive audiences away due to listening fatigue A PPM explanation and performance plots are available here http www indexcom com ppm Part 3 The Production Studio The role of the production studio varies widely from station to station If used only for creation of spots promos IDs etc production studio quality is considerably less critical than it is where programming is sweetened before being transferred to a playout system Our discussion focuses on the latter case Choosing Monitor Loudspeakers Th
110. p en wikipedia org wiki Flash_memory Data Compression Data compression is ubiquitous and choosing the correct compression algorithm codec is crucial to maintaining audio quality Almost all digital audio is delivered via some form of data compression algorithm Hence digital playout systems that use data compression should use the highest quality codec possible because the au dio will be compressed again at transmission Cascaded codecs can cause severe and unexpected loss of audio quality There are two forms of compression lossy and lossless Best modern practice is to use lossless or no data compression in an audio playout system Lossless Compression Lossless compression provides an output that is bit for bit identical to its input The only standards based lossless codec is MPEG 4 ALS formerly LPAC This has provi sions for tagging and metadata Some other lossless codecs include Windows Media Lossless used in Windows Media Player Apple Lossless used in QuickTime and Tunes FLAC Free Lossless Audio Codec WavPack and Shorten WinZip 11 0 and above uses WavPack to compress wav files and writes them to zipx format All of these algorithms remove statistical redundancy in the audio signal to achieve approximately 2 1 compression of audio that has not been heavily processed They have lower coding efficiency with material that has been subject to heavy dynamics compression and peak limiting like much of today s mus
111. ponents of a typical audio facility for operating level and headroom e Analog to digital converters e Studio to transmitter link land line microwave or optical fiber e Microphone preamps e Console summing amplifiers e Line amplifiers in consoles tape recorders etc e Distribution amplifiers if used e Signal processing devices such as equalizers e Specialized communications devices including remote broadcast links and telephone interface devices e Phono preamps e Tape and cart preamps e Record amplifiers in tape machines e Computer sound cards in the Broadcast Netcast Facility VU meters are worthless for checking peak levels Even peak program meters PPMs are insufficiently fast to indicate clipping of momentary peaks because their integra tion time is 5 or 10ms depending on which variant of the PPM standard is em ployed While PPMs are excellent for monitoring operating levels where small amounts of peak clipping are acceptable the peak signal path levels should be monitored with a true peak reading meter or oscilloscope Particularly if they are monitoring pre emphasized signals PPMs can under read the true peak levels by 5dB or more Adjust gains so that peak clipping never occurs under any reasonable operating conditions including sloppy gain riding by the operator It is important to understand that digital true peak reading meters also known as bit meters may show the peak value of digital sa
112. pressed files usually provide the highest quality For cuts that must be taken from vinyl disk it is preferable to use high end con sumer phono cartridges arms and turntables in production Make sure that one person has responsibility for production quality and for preventing abuse of the re cord playing equipment Having a single production director will also help achieve a consistent air sound an important contribution to the big time sound many sta tions want There are many low cost all digital mixers available Made by companies like Sound craft Yamaha Mackie and Roland these provide the ability to automate mixes and to keep the signal in the digital domain throughout the production process Although some people still swear by certain classic vacuum tube power amplifiers notably those manufactured by Marantz and McIntosh the best choice for a moni tor amplifier is probably a medium power 100 watts or so per channel solid state amplifier with a good record of reliability in professional applications 47 48 Maintaining Audio Quality Production Practices The following represents our opinions on production practices We are aware that some production facilities operate under substantially different philosophies But we feel that the recommendations below are rational and offer a good guide to achiev ing consistently high quality 1 Do not apply general audio processing to dubs and syndicated p
113. ps are notorious for low input overload points and can be easily clipped by high output microphones and or screaming announcers Similar problems can occur inside consoles if the console designer has poorly chosen gain structures and operating points or if the master gain controls are operated with unusually large amounts of attenuation When operating with nominal line levels of 4 or 8dBu the absolute clipping point of the line amplifier becomes critical The headroom between nominal line level and the amplifier clipping point should be greater than 16dB A line amplifier for a 4dBu line should therefore clip at 20dBu or above and an amplifier for a 8dBu line should clip at 24dBu or above IC based equipment which almost always clips at 20dBu or so unless transformer coupled is not suitable for use with 8dBu lines 4dBu lines have become standard in the recording industry and are preferred for all new studio construction recording or broadcast because of their compatibility with IC opamp operating levels 25 26 Maintaining Audio Quality The same headroom considerations that apply to analog also apply to many digital systems The only digital systems that are essentially immune to such problems are those that use floating point numbers to compute and distribute the digital data While floating point arithmetic is relatively common within digital signal processors mixers and digital audio workstations it is very
114. r comparing the CBS and BS 1770 meters http www orban com support orban techtopics White Paper BS 1770 vs CBS meter pdf Peak Normalization in Audio Editing Programs Many audio editing programs permit a sound file to be normalized which ampli fies or attenuates the level of the file to force the highest peak to reach 0 dBFS This is unwise for several reasons Peak levels have little to do with loudness so normal ized files are likely to have widely varying loudness levels depending on the typical peak to average ratio of the audio in the file Also if any processing occurs after the normalization process such as equalization one needs to ensure such processing does not clip the signal path If the processing adds level one must compensate by applying attenuation before the processing to avoid exceeding O dBFS or by using floating point arithmetic If attenuation is applied one must use care to ensure that the attenuated signal remains adequately dithered see page 37 Moreover normalization algorithms often do not use true peak level as specified in ITU Recommendation BS 1770 or oversampling If they do not files normalized by the algorithms can clip downstream D A and sample rate converters due to the OdBFS phenomenon see page 25 and cause more distortion and aliasing The audio processor analog input A D should clip at the same audio level as the source amplifier or console The input level should be adjusted so
115. r dynamics processing because of the excessive limiting and clipping applied to all too many of today s CDs Very experienced engineers master major label CDs using the best available processing and moni toring equipment typically costing over 100 000 per room in a well equipped mastering studio The sound of major label CDs represents an artful compromise between the demands of different types of playback systems and is designed to sound good on all of them although this goal is often compromised by today s CD loudness wars Mastering engineers do not make these compromises lightly We believe it is very unwise for a radio station to significantly depart from the spectral balance typical of major label CDs because this almost certainly guaran tees that there will be a class of receivers or players on which the audio sounds terrible 3 Mono Sources to Stereo Destinations When audio sources are mono it is best to take the mono signal and route it to both stereo channels This is particularly important with analog magnetic tape where phase error can be a problem If a mono tape is played using a stereo in the Broadcast Netcast Facility head use the best channel for both the left and right channels of the playout media This ensures that both stereo channels of the broadcast netcast will be identically mono Pay particular attention to the maintenance of production studio equip ment Even greater care than that employed in maintaining
116. radio Conversely the FM broadcast reception system can ex aggerate flaws in audio quality Audio processors even OPTIMOD are especially prone to exaggerating such flaws in the Broadcast Netcast Facility In this discussion we have tried to touch upon the basic issues and techniques un derlying audio quality in radio operations and to provide useful information for evaluating the cost effectiveness of equipment or techniques that are proposed to improve audio quality In particular we concluded that today s high quality IC opamps are ideally suited for use as amplification elements in broadcast and that compromises in digital standards computer sound cards disk playback and tape quality are all likely to be audible on the air The all digital signal path is probably the single most important quality improvement that a station can make but the in stalling engineer must be aware of issues such as lossy compression particularly when cascaded word length sample rate headroom jitter and dither and OdBFS induced clipping Following the suggestions presented here will result in better broadcast netcast au dio quality and that is a most important weapon in attracting and maintaining an audience that is routinely exposed to compact discs and other high quality audio reproduction media Provide your audience with the best possible experience The future belongs to the quality conscious 55 Maintaining Audio Quality
117. reo 19 20 Maintaining Audio Quality is optimized for the range of 16 40 kbps and provides high audio quality at bitrates as low as 24 kbps The Parametric Stereo encoder extracts a parametric representation of the stereo image of an audio signal Meanwhile a monophonic representation of the original signal is encoded via AAC SBR The stereo image information is represented as a small amount of high quality parametric stereo information and is transmitted along with the monaural signal in the bit stream The decoder uses the parametric stereo information to regenerate the stereo image This improves the compression effi ciency compared to a similar bit stream without Parametric Stereo MPEG Surround MPEG Surround standardized in ISO IEC 23003 1 2007 is an efficient technology for multi channel audio compression that extends the concept of parametric stereo to more than two channels Rather than performing a discrete coding of the individual audio input channels MPEG Surround captures the spatial image of a multi channel audio signal into a compact set of parameters that are used to synthesize a high quality multi channel representation from a transmitted downmix signal MPEG Surround combines a core audio codec usually AAC or HE AAC although other codecs like MP3 can be used with a parametric side channel containing the information necessary to distribute the audio to the various surround output chan nels Typically t
118. rformance at 192 kbps and higher is still very good Spectral Band Replication Low bitrate audio coding is an enabling technology for a number of applications like digital radio Internet streaming netcasting webcasting and mobile multimedia applications The limited overall bandwidth available for these systems makes it nec essary to use a low bitrate highly efficient perceptual audio codec in order to create audio that will attract and hold listeners In Internet streaming applications the connection bandwidth that can be estab lished between the streaming server and the listener s client player application de pends on the listener s connection to the Internet In many cases today people use analog modems or ISDN lines with a limited data rate lower than the rate that can produce appealing audio quality with conventional perceptual audio codecs Moreover even if consumers connect to the Internet through high bandwidth con nections such as xDSL or CATV the ever present congestion on the Internet limits the connection bitrate that can be used without audio dropouts and rebuffering Furthermore when netcasters pay for bandwidth by the bit using a highly efficient perceptual codec at low bitrates can make netcasting profitable for the first time In mobile communications the overall bandwidth available for all services in a cer tain given geographic area a network cell is limited so the system operator must take measures to all
119. rick Emil L A New Loudness Indicator for Use in Broadcast ing J SMPTE September 1981 pp 772 777 in the Broadcast Netcast Facility ment IEEE Transactions on Audio and Electroacoustics Volume AU 14 Number 3 September 1966 pp 141 151 This paper also presented results from other tests whose goal was to model the loudness integration time constants of human hear ing BS 1770 Loudness Meter In 2006 the ITU R published Recommendation ITU R BS 1770 Algorithms to meas ure audio programme loudness and true peak audio level In 2011 this was up dated to BS 1770 3 which adds gating so that the meter ignores silence and is weighted toward louder program material which contributes most to a listener s perception of loudness BS 1770 3 indicates only sounds that fall within a floating window that extends from the loudest sounds within the preset integration period to sounds that are 10 dB quieter than the loudest sounds Developed by G A Soulodre the original BS 1770 loudness meter uses a frequency weighted RMS measurement intended to be integrated over several seconds per haps as long as an entire program segment As such it is considered a long term loudness measurement because it does not take into account the loudness integra tion time constants of human hearing as does the CBS meter A major disadvantage of the BS 1770 1 meter is that it weights silence and low loudness material the same as hi
120. rograms from commercial recordings in the production studio OPTIMOD provides all the processing necessary and does so with a remarkable lack of audible side effects Further compression is not only undesirable but is likely to be very audible If the production compressor has a slow attack time and therefore produces overshoots that can activate gain reduction in OPTIMOD it will probably fight with a downstream OPTIMOD ultimately yielding a substantially worse air sound than one might expect given the individ ual sounds of the two units If it proves impossible to train production personnel to record with the correct levels we recommend using the Orban Optimod PC to protect the production recorder from overload When used for leveling only Optimod PC does not af fect short term peak to average ratio of the audio and so will not introduce un natural artifacts into OPTIMOD processing Optimod PC is an audio processor on a sound card and can be used in any Windows XP or Vista computer such as the one that may already be present in the production studio 2 Avoid excessive bass and treble boost Substandard recordings can be sweetened with equalization to achieve a tonal balance typical of the best currently produced recordings However avoid exces sive treble boost because it will stress AM and FM audio processors We recom mend using a modern CD typical of your program material as a reference for spectral balance although not fo
121. s and tape irregularity are also reduced and the effects of interchannel phase shifts are halved However a playback machine can deteriorate due to oxide build up on the heads or incorrect azimuth far more severely at 15ips than at 7 5ips before an audible change occurs in audio quality Noise Reduction A compander type encode decode noise reduction system can be used to reduce tape hiss to an unobtrusive or even inaudible level However if transparency is de sired it is difficult to imagine a contemporary broadcast application where com panded analog tape would be preferred to linear PCM digital recording which is reliably transparent when implemented correctly In contemporary production tape is usually used because it colors the sound in a way that artists and producers find attractive Tape hiss soft saturation and modulation noise are part of that color Because 7 5 ips introduces more high frequency saturation than 15 ips the produc tion community has found both speeds to be useful for different effects However compander technology is still of interest because many legacy recordings were recorded using a compander type noise reduction system and correct playback requires access to the compander hardware or a digital model of it We have evalu ated and can enthusiastically recommend Dolby SR Spectral Recording Good re sults have been reported with Telcom C4 as well dbx Type II noise reduction is also effective and has th
122. s cannot Because of the lower reflectivity the lasers within broadcast audio CD players need to be in good condition to read CD R without errors Sometimes all that is necessary is a simple cleaning of the lens to restore satisfactory performance CD RW compact disk rewritable is not a true random access medium You cannot randomly erase cuts and replace them because the cuts have to be unfragmented and sequential However you can erase blocks of cuts always starting backwards with the last one previously recorded You can then re record over the space you have freed up The disadvantage of CD RW is that some CD payers cannot read them unlike CD R which can be read by almost any conventional CD player if the disk has been final ized to record a final Table of Contents track on it A finalized CD R looks to any CD player like an ordinary CD Once a CD R has been finalized no further material can be added to it even if the disk is not full If a CD R has not been finalized it can only be played in a CD R recorder or in certain CD players that specifically support the playing of unfinalized CD Rs M DISC M DISC sells archival media using a stone like formulation that according to the company provide a 1000 year archival lifetime These are available in DVD and Blu ray formats They can be read by any DVD or Blu ray reader but must be burned with M DISC compatible writers which are available from several major manufac turers A
123. s visi ble If the cartridge mistracks asymmetrically indicating incorrect anti skating compensation then the scope trace will be bent at its ends If this happens adjust the anti skating until the trace is a straight line indicating symmetrical clipping It is important to note that in live disk operations use of anti skating compensa tion may increase the chance of the phono arm sticking in damaged grooves in stead of jumping over the bad spots Increasing tracking force by approximately 15 has the same effect on distortion as applying anti skating compensation This alternative is recommended in live disk operations Use a modern direct drive turntable None of the older types of professional broadcast turntables have low enough rumble to be inaudible for broadcast or netcast These old puck belt or gear driven turntables might as well be thrown away Multiband audio process ing can exaggerate rumble to extremely offensive levels Mount the turntable properly Proper turntable mounting is crucial an improperly mounted turntable can pick up footsteps or other building vibrations as well as acoustic feedback from monitor speakers which will cause muddiness and severe loss of definition The turntable is best mounted on a vibration isolator placed on a non resonant ped estal anchored as solidly as possible to the building or preferably to a concrete slab The turntable bases supplied by the turntable manufact
124. s with antenna bandwidth and group delay can also cause synchronous FM as can excessive VSWR which causes reflections to occur between transmitter and antenna Perhaps the most severe antenna induced problems relate to coverage pattern Proper choice of the antenna and its correct installation can dramatically affect the amount of multipath distortion experienced by the listener Multipath induced de gradations are far more severe than any of the other quality degrading factors dis cussed in this paper Minimization of received multipath is the single most important thing that the broadcast engineer can do to ensure high quality at the receiver AM Transmitter We live in the golden age of AM transmitters After 75 years of development we finally have AM transmitters using digital modulation technology that are audibly transparent even at high power levels Previously even the best high power AM transmitters had a sound of their own and all audibly degraded the quality of their inputs We recommend that any AM station that is serious about quality upgrade to such a transmitter By comparison to any tube type transmitter not only is the quality au dibly better on typical consumer receivers but the transmitter will pay for itself with lower power bills 15 Geoff Mendenhall of Harris has written an excellent practically oriented paper on minimiz ing synchronous FM G Mendenhall Techniques for Measuring Synchronous FM Noise in FM
125. sed on a case by case ba sis if nothing better is available If you use sources with lossy encoding like AAC be sure to use a decoder that uses floating point arithmetic or some other mechanism that prevents codec induced overshoots from clipping the audio Fixed point decoding is acceptable if the de coder s designer has built adequate headroom into the decoder Many original record labels are defunct and have transferred licenses to other la bels It was formerly safe to assume that the audio from the original record CD label or authorized licensee is as good as it gets but tasteless remastering has ruined many recent major label re releases even within the same labels Many major labels produce collections for other well known marketing groups Many of these sources are acceptable although they require careful auditioning and quality validation Some smaller and obscure labels have acquired licenses from the original labels While some of this work has proven to be excellent some of these reissues should probably be avoided Syndicated programming can be another source of audio quality problems espe cially if the audio source material has been preprocessed in any way Such program ming should be very carefully validated Another pitfall in CD reissues is mono compatibility This is important because most clock radios and mobile phone speakers are mono and every FM car radio imple ments blend to mono as a function of signal strength
126. sic Balance lA 33 ELCCUONICOUGLIOY eno ot MII BE MA ait MEI AIA EAEE MAI nt MUN AL 34 There are several pervasive myths regarding digital audio ooooononocinoniconononononnconocononononncononnncnnonnnnns 38 And finally some truisms regarding loudness and qualitY oocoonoccicciccnoccnoconcnonncononononccononan cn nono nonncnnno 40 The Arbitron PPM Portable People Meter PART 3 TBE PRODUCTION STUDIO aeni e a e a suesneccspeetnsdbvagnevesassepoceeveenens Choosing Monitor Loudspeakers Loudspeaker Location and Room Acoustics Loudspeaker Equalization Stereo Enhancement 0c00000 Other Production Equipment Production Practices condal elias PART 4 EQUIPMENT FOLLOWING OPTIMOD AMET E MET VGNSMUCP a it da tedio laa ceda della axe EMANAN a a ese AM Transmitter AM Antenna osooso DAB HD Radio Netcasting Encoders Audio Processing for Low Bitrate Digital Transmissions SUMMARY a NN APPENDIX ANALOG MEDIA vieron e E E EN GEA T caida Authors Note for the 2014 Edition resore aia E aE EK E AE KEE AEA 56 VEDIA A NEARE E A EAEE A E E AS Analog LADEN A A A A Sum and Difference Recording Electro ic Phase Corrections radios add CAM lid 62 ES 20s 1 8 sk os sas etactes Bete cs sole Soaneties T A EEE AAE es an sca AEEA 63 NOE 63 Tape Recorder Maintenance cccccccccccececssesetseesecnseeecusesseesecseesecseeecseesenseeseeseeseseeeeeneeeeeaaeats 64 Recording Your Own Alignment Tapes cc cccccccc
127. studies that led to his EE degree he took the first three years of the composition and music theory courses usually taken by music majors Orban has been involved in professional recording for many years In Princeton he supplemented his income by offering on location recording services recording many performances for both for the 17 kW commercially licensed campus radio station WPRB FM and for the University itself He was closely associated with WPRB in the Broadcast Netcast Facility throughout his college years hosting a weekly classical music show and serving at various times as Chief Engineer and Music Director He also became a skilled top 40 board operator when this meant relying on live talent cart machines and slip cued vinyl It was at WPRB that he designed and built his first audio processor and caught the radio bug that led to his career as a successful broadcast equipment designer Around 1970 he became associated with electronic music pioneers Paul Beaver and Bernie Krause and mixed several of their records for the Warner Bros label Later he worked with a number of independent artists and labels as a writer mixer in strumentalist and producer Orban s compositions been heard on classical radio sta tions in New York and San Francisco and his score for a short film Dead Pan was heard on PBS television in Chicago He was able to exploit his experience in pro au dio when designing studio reverberators stereo
128. the noise print and to dynamically update it throughout the program being treated These automatic systems are particularly valuable for vinyl noise reduction where unlike analog tape the noise floor is unlikely to be statistically stationary At the high end the line of hardware based processors made by CEDAR in England has established itself as being the quality reference for this kind of proc essing The CEDAR line is however very expensive by comparison to the plug ins described above 17 http www diamondcut com 18 http www izotope com 19 http www cedar audio com in the Broadcast Netcast Facility Other high end products include the Sonic Solutions No Noise system available as part of the Sonic Solutions workstations for mastering applications and the TC Restoration Suite for the Powercore Platform Be aware that not all noise reduction systems or algorithms perform equally well Some are dramatically more effective than others and some compromise audio quality more than others Unsophisticated algorithms can suppress transients distort brass and string sounds and sound flangy like MP3 Analog Tape Despite its undeniable convenience the tape cartridge even at the current state of the art is inferior to reel to reel in almost every performance aspect Performance differences between cart and reel are readily measured and include differences in frequency response noise high frequency headroom wo
129. the point being re constructed The more samples that are used the better and more accurate the re sult even if this means that the filter is very long It s easiest to justify this assertion in the frequency domain Provided that the fre quencies in the passband and the transition region of the original anti aliasing filter are entirely within the passband of the reconstruction filter then the reconstruction filter will act only as a delay line and will pass the audio without distortion Of course all practical reconstruction filters have slight frequency response ripples in their passbands and these can affect the sound by making the amplitude response but not the phase response of the delay line slightly imperfect But typically these ripples are in the order of a few thousandths of a dB in high quality equip ment and are very unlikely to be audible The authors have proved this experimentally by simulating such a system and sub tracting the output of the reconstruction filter from its input to determine what er rors the reconstruction filter introduces Of course you have to add a time delay to the input to compensate for the reconstruction filter s delay The source signal was random noise applied to a very sharp filter that band limited the white noise so that its energy was entirely within the passband of the reconstruction filter We used a very high quality linear phase FIR reconstruction filter and ran the simulation
130. this clip level is the same Replay Gain A popular means of estimating and controlling the loudness of audio files is the Re play Gain technique The computes a gain factor to be applied to the file when played back this gain factor is stored as metadata in the file header The goal is to achieve consistent long term loudness from track to track The gain factor is com puted by the following steps 1 Equal Loudness Filtering The human ear does not perceive sounds of all frequencies as having equal loud ness For example a full scale sine wave at 1kHz sounds much louder than a full scale sine wave at 10kHz even though the two have identical energy To account for this the signal is filtered by an inverted approximation to the equal loudness curves sometimes referred to as Fletcher Munson curves 2 RMS Energy Calculation Next the energy during each moment of the signal is determined by calculating the Root Mean Square of the waveform every 50ms 3 Statistical Processing 8 http replaygain hydrogenaudio org index html in the Broadcast Netcast Facility Where the average energy level of a signal varies with time the louder moments contribute most to our perception of overall loudness For example in human speech over half the time is silence but this does not affect the perceived loudness of the talker at all For this reason the RMS values are sorted into numerical order and the value 5 down the list is chosen
131. tightly so they can enjoy television programs at low volumes and or in noisy environments Producing more dynamic range com pression without objectionable side effects requires use of multiband compressors like that found in Optimods designed for digital television such as Optimod Surround 8685 When AC3 is used as a contribution codec it is possible to configure the Dolby AC3 decoder incorrectly so that it applies unwanted and unexpected wideband compres sion to its output If your facility is using AC3 as a contribution codec it is important to double check the configuration of the decoder to make sure that dynamic range control is not applied to the decoder s output It is safest to choose a DRC profile of None at the AC3 encoder This ensures that dynamic compression cannot be acci dentally applied to the signal in the decoder Part 2 System Considerations Wiring and Grounding Analog Interconnection For analog connections we recommend balanced connections between devices us ing XLR type connectors for termination because of their robustness Use two conductor foil shielded cable such as Belden 8451 1503A 1504A 1508A or equiva lent because signal current flows through the two conductors only The shield does not carry signal and is used only for shielding It should be connected at the input only to prevent ground loop hum Digital Interconnection Per the AES3 standard each digital input or output line carries
132. track Striking the top in the Broadcast Netcast Facility and sides of the player with varying degrees of force while listening to the output can give a feel for the player s vibration resistance Fortunately some of the play ers with the best sound also track best The depressing trade off between quality and ruggedness that is inevitable in vinyl disk reproduction is unnecessary when CDs are used Reliability is not easy to assess without experience The experience of your fellow broadcasters can be valuable here ask around during local broadcast engineers meetings Be skeptical if examination of the insides of the machine reveals evi dence of poor construction Cueing and interface to the rest of the station are uniquely important in broadcast There are at this writing relatively few players that are specifically designed for broadcast use players that can be cued by ear to the start of a desired selection paused and then started by a contact closure The practical operation of the CD player in your studio should be carefully considered Relatively few listeners will no tice the finest sound but all listeners will notice miscues dead air and other obvious embarrassments Some innovative designs that have already been introduced include jukebox like CD players that can hold 100 or more CDs These players feature musical selections that can be chosen through computer controlled commands An alternative design
133. ty in a given brand of ma chine Run tests on various brands of carts and standardize on the one offering best phase stability 3 Follow the manufacturer s maintenance and alignment instructions Because of the vast differences in design from manufacturer to manufacturer it is difficult to provide advice that is more specific 4 Consider upgrading the cart machine s electronics Many early and some not so early cart machines had completely inadequate electronics The performance of these machines can be improved considerably by certain electronics modifications Check the machine for the following 24 Useful discussions of sticky shed syndrome can be found at http Awww clir org pubs reports pub54 2what_wrong html and http mixonline com ar audio sleep egyptian Maintaining Audio Quality A record amplifier headroom be sure the amplifier can completely saturate the tape before it clips B record amplifier noise and equalization some record amplifiers can actually contribute enough noise to dominate the overall noise performance of the machine C playback preamp noise and compliance with NAB IEC equalization D power supply regulation noise and ripple E line amplifier headroom F record level meter alignment to improve apparent signal to noise ratio at the expense of distortion some meters are calibrated so that 0 corresponds to sig nificantly more than 1 third harmonic distortion Probably the most
134. uch codecs should not use clippers for limiting and should instead use a look ahead type limiter see page 53 OPTIMOD SURROUND OPTIMOD DAB OPTIMOD HD FM and OPTIMOD PC provide the correct form of peak limiting for these applications and other low bite rate digital audio services Just as the motion picture industry creates a consistent professional look to their product by applying exposure and color correction to every scene in a movie audio processing should be used as part of the audio broadcast product to give it that final professional polish Achieving consistent state of the art audio quality in broadcast is a challenging task It begins with a professional attitude considerable skill patience and an unshak able belief that quality is well worth having It usually requires the careful coopera in the Broadcast Netcast Facility tion between programming engineering and computer IT departments With the advent of computer based audio systems and computer network delivered audio it requires extra computer IT knowledge Computer IT personnel should understand digital audio fundamentals This document provides some technical insights and tips on how to achieve immaculate audio and keep it that way Remember successful audio processing results all start at the source This publication is organized into four main parts 1 Recording media Compact disc CD R and CR RW DVD R DVD RW DVD A HD DVD Blu ray digital tape m
135. ue to tilt and bounce to less than 1 modulation Therefore either technology can provide excel lent results FM Transmitter The transmitter must be transparent to the modulated RF If its amplifiers are nar rowband lt 500 kHz at the 3dB points it can significantly truncate the Bessel side bands produced by the FM modulation process introducing distortion For best re sults 3dB bandwidth should be at least 1MHz 4 Co author Greg Ogonowski Orban s Vice President of New Product Development origi nally brought this to the industry s attention www indexcom com Ogonowski has devel oped modifications for several exciters and STLs that improve the transient response of their AFCs 51 52 Maintaining Audio Quality Narrowband amplifiers can also introduce synchronous FM This can cause audible problems quite similar to multipath distortion and can particularly damage SCAs Synchronous FM should be at least 35dB below carrier level with 40dB or better preferred 5 If the transmitter s group delay is not constant with frequency it can also introduce synchronous FM even if the bandwidth is wide Please note that the Incidental FM reading on most FM modulation monitors is heavily smoothed and de emphasized and cannot be used to measure synchronous FM accurately At least one device has appeared to do this accurately Radio Design Labs Amplitude Com ponent Monitor Model ACM 1 FM Antenna Problem
136. um SBR reconstructs the higher frequencies in the decoder based on an analysis of the lower frequencies transmitted in the underlying coder To ensure an accurate recon struction some guidance information is transmitted in the encoded bitstream at a very low data rate The reconstruction is efficient for harmonic as well as for noise like components and permits proper shaping in both the time and frequency domains As a result SBR allows full bandwidth audio coding at very low data rates and offers significantly increased compression efficiency compared to the core coder SBR can enhance the efficiency of perceptual audio codecs by 30 even more in certain configurations in the medium to low bitrate range The exact amount of improvement that SBR can offer also depends on the underlying codec For instance combining SBR with AAC achieves a 64 kbps stereo stream whose quality is compa rable to AAC at 96 kbps stereo SBR can be used with mono and stereo as well as with multichannel audio SBR offers maximum efficiency in the bitrate range where the underlying codec it self is able to encode audio signals with an acceptable level of coding artifacts at a limited audio bandwidth Parametric Stereo Parametric Stereo is the next major technology to enhance the efficiency of audio compression for low bitrate stereo signals Parametric Stereo is fully standardized in MPEG 4 and is the new component within HE AACv2 As of today Parametric Ste
137. urer are highly rec ommended Use a properly adjusted high quality phono preamp Until recently most professional phono preamps were seriously deficient com pared to the best high end consumer preamps Fortunately this situation has changed and a small number of high quality professional preamps are now available mostly from small domestic manufacturers A good preamp is charac terized by extremely accurate RIAA equalization high input overload point bet ter than 100mV at 1 kHz low noise optimized for the reactive source imped ance of a real cartridge low distortion particularly CCIF difference frequency IM load resistance and capacitance that can be adjusted for a given cartridge and cable capacitance and effective RFI suppression The Firestone Audio Korora RIAA Phono Stage Preamp and SUPPLIER Power Supply Unit available from http soundadditions com are highly recommended After the preamp has been chosen and installed the entire vinyl disk playback system should be checked with a reliable test record for compliance with the RIAA equalization curve If you wish to equalize the station s air sound to pro in the Broadcast Netcast Facility duce a certain sound signature the phono preamp is not the place to do it Some of the better preamps have adjustable equalizers to compensate for fre quency response irregularities in phono cartridges Since critical listeners can de tect deviations of 0 5dB
138. ven standards based algorithm for example MPEG1 Layer 2 or AAC have equal performance However this is not true in prac tice Codec standards emphasize standardizing the decoders while allowing the en coders to be improved over time While it is expected that not all manufacturer s encoders will perform equally to a less extent this is also true of decoders Not every decoder realizes the standard in an ideal way for example there can be compro mises caused by using fixed point arithmetic in a codec whose reference code was implemented in floating point There can also be numeric inaccuracies caused by the sample rate conversion algorithms that are often included in the codec implementa tion Not all codecs of the same type have equal performance To assess the audible transparency of codecs the ITU has published Recommenda tion ITU R BS 1116 1 which is intended for use in the assessment of systems that in troduce impairments so small as to be undetectable without rigorous control of the experimental conditions and appropriate statistical analysis All of the high quality MPEG standard based codecs have been assessed using this algorithm and the re sults have been published Similarly the ITU has developed the BS 1534 1 standard commonly known as MUSHRA MUItiple Stimulus with Hidden Reference and Anchors which is widely used for the evaluation of systems exhibiting intermediate quality levels in particu lar low bitrate codecs MPE
139. versed TI has ac knowledged the problem and has released revised parts However many audio in terfaces and codecs in use have this problem and should be scrupulously avoided Level metering in sound cards is highly variable Average quasi peak and peak re sponses are all common and often inadequately or incorrectly documented see Metering on page 29 This is relevant to the question of line up level EBU R68 specifies reference level as 18dBfs while SMPTE RP 155 specifies it as 20dBfs Unless the sound card s metering is accurate it is impossible to ensure compliance with the standards maintained within your facility Many professional sound cards have adequate metering while this is far less common on consumer sound cards Further consumer sound cards often cannot accommodate professional analog lev els balanced lines or AES EBU inputs and outputs Bit Accuracy Digital systems having no sample rate converters in their signal path are considered synchronous if there is no further digital signal processing A master clock signal typically in wordclock or AES11 format is required for all devices in the audio signal path to remain in sync To be bit accurate a system must be synchronous and must not change the word length of the audio data being conveyed Digital systems having sample rate converters in their signal path are not bit accurate and are considered asynchronous even though they can also be locked to a master clock
140. w and flutter and particu larly azimuth and interchannel phasing stability Cassettes are sometimes promoted as a serious broadcast program source We feel that cassettes low speed tiny track width sensitivity to dirt and tape defects and substantial high frequency headroom limitations make such proposals totally im practical where consistent quality is demanded Sum and Difference Recording Because it is vital in stereo FM broadcast to maintain mono compatibility sum and difference recording is preferred in either reel or cart operations This means that the mono sum signal L R is recorded on one track and the stereo difference signal L R is recorded on the other track A matrix circuit restores L and R upon playback In this system interchannel phase errors cause frequency dependent stereo field localization errors rather than deterioration of the frequency response of the mono sum Because this technique tends to degrade signal to noise L R usually dominates forcing the L R track to be under recorded thereby losing up to 6dB of signal to noise ratio it is important to use a compander type noise reduction system if sum and difference operation is employed Electronic Phase Correction Because interchannel phase errors are endemic on analog tape it is wise to maintain a transfer machine in which the reproduce head azimuth adjustment is readily avail able for tweaking by ear This is particularly effective if the technicia
141. why Although the radio broadcast community has used processing before transmission for most of the history of the medium this has been thoroughly researched over the years to discover how it affects audiences In general the type of processing used on typical CDs these days has been shown to increase cume the number of distinct persons listening to a given station or stream in a one week period while driving average time spent listening down Moreover the kind of brightness present on many of the today s CDs has been shown by broadcasters to repel women listeners It is probably no coincidence that most mastering engineers are male The theory If a little sounds good a lot must be better usually does not apply to audio signal processing Meanwhile as CDs became more and more overprocessed their sales declined pre cipitously We suspect that illegal file sharing was not the only cause We have to wonder why the executives who run the labels refuse to make the connection be tween the sort of brutal overprocessing on many of today s CDs and the increasing lack of satisfaction with the product It s not as if they haven t been told Here are some interesting references on the CD Loudness Wars http www npr org templates story story php storyld 122114058 http flowingdata com 2010 01 05 a visual history of loudness in popular music CD R and CD RW DVD R DVD RW DVD A HD DVD Blu ray Recordable optical media ar
142. with better transient response and lower low frequency distortion At this writing the state of the art in analog modulated oscillator distortion is ap proximately 0 02 THD at 75 kHz deviation Distortion in digital exciters is typi cally 10 times lower than this In our opinion if the THD of your exciter is less than 0 1 it is probably adequate If it is poorer than this as many of the older technol ogy exciters are replacing your exciter will audibly improve sonic clarity and will also improve the performance of any subcarriers Even if the distortion of your modulated oscillator is sufficient the performance of the AFC loop may not be A high performance exciter must have a dual time constant AFC loop to achieve satisfactory low frequency performance If the AFC uses a compromise single time constant stereo separation and distortion will be compromised at low frequencies Further the exciter will probably not accurately reproduce the shape of the carefully peak controlled OPTIMOD FM output intro ducing spurious peaks and reducing achievable loudness Even dual time constant AFC loops may have problems If the loop exhibits a peak in its frequency response at subsonic frequencies it is likely to bounce and cause loss of peak control Composite STLs can have similar problems Digital exciters have none of these problems However a properly designed analog exciter can have good enough performance to limit overshoot d
143. xed amount usually 1 5 to 3dB the correct amount of decrease is a function of both tape formulation and the width of the gap in the record head consult the tape manufacturer s data sheet 9 Follow the manufacturer s current recommendations In addition to the steps listed above most tape machines require periodic brake adjustments reel holdback tension checks and lubrication With time critical bearings will wear out in the motors and elsewhere such failures are usually in dicated by incorrect speed increased flutter and or audible increases in the me chanical noise made by the tape recorder Use only lubricants and parts specified by the manufacturer 10 Keep the tape recorder and its environment clean Minimize the amount of dust dirt and even cigarette smoke that comes in con tact with the precision mechanical parts In addition to keeping dust away from the heads and guides periodically clean the rest of the machine with a vacuum cleaner in suction mode please or with a soft clean paintbrush It helps to re place the filters in your ventilation system at least five times per year Recording Your Own Alignment Tapes Recording a secondary standard alignment tape requires considerable care We recommend you use the traditional series of discrete tones to make your secon dary standard tapes A Using a standard commercial alignment tape very carefully align the playback section of the master recorder on which the
144. y systems which relate more to ergonomics and reliability than to audio quality How ever two crucial issues are how the audio is input and output from the system and whether the audio data is stored in uncompressed linear PCM form or using some sort of data compression Audio is usually input and output from these systems through sound cards Please see the discussion on page 26 regarding sound cards and line up levels 12 Maintaining Audio Quality Flash RAM As its price continues to fall flash RAM has become ever more popular as an audio storage medium It is available packaged with a controller with many different in terfaces USB2 0 being one of the most popular and universally compatible Unlike CD RW and DVD RW flash RAM is capable of tens of thousands of writes and hun dreds of thousands of reads It can be written to and read from in faster than real time for any commonly used audio sample rate and bit depth Flash RAM is available in two main technologies called NOR and NAND types Both are usable for audio recording and have different trade offs While neither is capa ble of true random access writes because old data must be block erased this is usu ally not a limitation for audio recording The long term storage reliability of flash RAM has not yet been proven and it is therefore unwise to rely on flash RAM as a sole means for archival backup This article provides a good summary of flash RAM technology htt
145. y of any audio codec AAC was chosen as the core codec for HE AAC because of its superior performance over older generation audio codecs such as MP3 or WMA This was one of the main reasons why Apple Computer chose AAC for their market dominating iTunes downloadable music service HE AAC delivers streaming and downloadable audio files at 48 kbps for FM quality stereo entertainment quality stereo at 32 kbps and good quality for mixed content even below 16 kbps mono This efficiency makes new applications in the Internet mobile and digital broadcast markets viable Moreover unlike certain other pro prietary codecs AAC HE AAC does not require proprietary servers for streaming AAC HE AAC can be stream delivered using SHOUTcast Icecast2 QuickTime Darwin Real Helix Adobe Flash and Wowza Media Servers Members of the HE AAC Codec Family HE AACv1 combines AAC and SBR HE AACv2 builds on the success of HE AACv1 and adds more value where highest compression efficiency for stereo signals is re 3 http en wikipedia org wiki Advanced_Audio_Coding 15 16 Maintaining Audio Quality quired HE AACv2 is a true superset of HE AACv1 as HE AACv1 is of AAC HE AACv2 adds Parametric Stereo to HE AACv1 further improving coding efficiency at low bi trates The members of the HE AAC codec family are designed for forward and backward compatibility Besides HE AACv2 bit streams an HE AACv2 encoder is also capable of creating HE AACv1 an
146. y thing that raising the sample frequency does is to add ultrasonic images of the original audio around the new sample frequency In any correctly designed sam ple rate converter these are reduced but never entirely eliminated by a filter fol lowing the upsampler People who claim to hear differences between upsampled audio and the original are either imagining things or hearing coloration caused by the added image frequencies or the frequency response of the upsampler s filter They are not hearing a more accurate reproduction of the original recording This also applies to the sample rate conversion that often occurs in a digital facil ity It is quite possible to create a sample rate converter whose filters are poor enough to make images audible One should test any sample rate converter hard ware or software intended for use in professional audio by converting the highest frequency sinewave in the bandpass of the audio being converted which is typically about 0 45 times the sample frequency Observe the output of the SRC on a spec trum analyzer or with software containing an FFT analyzer like Adobe Audition In a professional quality SRC images will be at least 90 dB below the desired signal and in SRC s designed to accommodate long word lengths like 24 bit images will often be 120 dB or lower A good reference on sample rate conversion perform ance can be found here http src infinitewave ca And finally some truis

Maintaining Audio Quality in the Broadcast/Netcast Facility

Contents

Download Pdf Manuals

Related Search

Related Contents