A standard music CD stores musical sounds as a sequence of 16-bit numbers with 44.1 kHz sampling frequency. The sound record with such parameters does not meet the abilities of a human hearing sense. So the quality of a sound seems to be not perfect. Especially, if one can compare a CD record with the original music recorded in 20 or 24-bit format with 96 kHz sampling frequency on an expensive studio hardware. Studio engineers try to explain sound deterioration in a CD format using inexplicable words such as "dead sound", "murder of a sound", "dimness", "lack of transparence" and even "low resolution". Thus they do make impression on credulous musicians.
Let us have a close look at a CD recording. Usually it takes place in a small sound absorbing room. As a rule a sound goes to microphones and then to a mixing console. Some analog devices can get worse the quality of the sound even before its analog-digital processing. So it is recommended to avoid condenser microphones with phantom power supply. If you can not afford a professional mixing console (the price can be up to $25000) you should first make analog-digital conversion by 20..24-bit computer sound cards. Then you can perform all the mixing operations by digital signal processing hardware or software. It is better to record digital sound data straight to the hard disc. Using DAT type recorders might bring about the jitter problems.
CD creation software
All the processing of a digital audio record can be fully done by special software. The capabilities of the modern professional quality software packages based on 24..32-bit float point mathematics exceed that of studio equipment. Furthermore, the whole processing can be realized in digital domain. It eliminates sound distortion typical for studio reverbs and other equipment without digital inputs and outputs like Yamaha SPX90, Sony DPS-R7, Alesis MidiVerb, DOD R-512 etc. When you use such devises an analog signal should be turned in digital data (not always 24-bit), then processed by DSP built in reverbs gear (you can not always expect it uses 32-bit floating point math), turned back to analog (as a rule without Noise Saping or Dithering). After that the signal goes to a mixing board and then to analog to digital converter of a studio digital type recorder. Naturally such transformations do not guaranty sound quality and bring about legends that digit murders the sound. So, it is very wise to stick to totally digital sound recording on PC applying at least 24-bit software for sound processing, mixing and mastering. See below the characteristics of some major products of the software developers.
SAW by
Innovative Quality Software grants up to 24 tracks for recording, playing back and sound processing. It enables independent real time (for some effects) processing of recorded tracks by reverbs, chorus, flanger, multi-channel compression, noise suppression, phase inversion, graphical equalizer, echo delay and Doppler effects. SAW can support simultaneously up to 8 sound cards. The value of a sampling frequency is determined for each track separately. So, it makes possible to record files with various sampling frequencies in a single project. Mixing properties provide the easy creation of a stereo record. Besides Innovative Quality Software offers different plug-ins such as MultiTrack View, PanHandler, AVI Viewer, Ignition CD. They extend considerably the field of SAW applications. Thus if you have got a CD-RW then you can make a CD record using SAW plus IgnitionCD plug-in. I should mention that Noise Shaping and Dithering processing modules are absent in standard SAW shipping. It is its fault.Sound Forge 4.5 by
Sonic Foundry is designed as a 32-bit multifunctional software for professional editing of audio files. Plug-ins by Sonic Foundry, Steinberg and Waves multiply the number of its functions. Very efficient algorithms allow real time sound processing. User's interface provides easy access to all the functions. Sound Forge manipulates with audio files of different formats and uses various algorithms of data compression (RealAudio, ADPCM, MPEG, MP3). Note that it edits the copy of an audio file. Only after you are fully happy with the result of the editing you can save all the corrections to the original file.Sound Forge enables applying of reverbs, chorus, harmonizer, sound compressor and expander, noise suppression, parameters and 10-channel graphical equalizer. Multi-channel compressor efficiently abridges dynamic range of a signal in particular frequency bands. It eliminates the influence of the powerful low frequencies on the middle frequency diapason during signal compression. Sound Forge can help in the restoration of old vinyl records. Firstly, Noise Reduction plug-in measures the noise level in the source audio record. Then it does some calculations to understand the nature of the noise. And finally, Noise Reduction plug-in removes intelligently the noise from the whole record. Additionally, Noise Reduction plug-in has a special tool to eliminate sound clicks using an interpolation method. Sound Forge has a Spectrum Analysis plug-in. It shows the spectrum of either the whole file or its selected part. It enables real time spectrum monitoring during playback. Sound Forge can synthesize the following wave forms - sine, meander, tooth with predefined amplitude and frequency. It can generate white noise as well. Sound Forge can imitate the sounds of analog synthesizers. Loop Tuner function helps to create and edit cycling samples. Sound Forge provides editing of the audio tracks built in AVI-files. This is a useful feature in the multi-media production.
It is impossible to do without MIDI waves table synthesizers.
Digital Orcestrator by Steinberg and Cakewalk by Twelve Tone are the most famous sequencers. These software packages work as virtual recording studios. MIDI wave table voices are recorded with the virtual multi-track type recorder. Note that you are able to record a track and play back the other ones simultaneously. "Computer" composers can record different instruments to different tracks from a MIDI-clavier, microphone or line inputs, fix the sound volume separately for each instrument and make sound processing with DSP. DSP is installed on the most widespread sound cards such as SBLive, AWE32/64, Turtle Beach Pinnacle, Monster Sound 3D, Ensoniq or YAMAHA DB50/60X. Then the tracks are mixed, processed and recorded as a usual stereo WAV-file. It is also possible to export all the tracks to WAV-files. And after that you can use SAW software to make a stereo sound, compression and sound processing. As a result you can get the computer imitation of an orchestra sound more or less closely.WaveLab 3.0 is a perfect software for sound recording, editing, processing and subsequent CD recording. It enables real time sound processing simultaneously applying up to 6 digital effects. WabeLab makes use of all Windows 95/NT achievements in 32-bit multi-task technology. While some WAV-files are playing back the other ones can be downloaded, processed and saved. WaveLab provides 24-bit data processing with dynamic scope up to 192 dB and signal/noise ration up to 144 dB. At last it has a built-in function of CD producing. Moreover all the music CDs agree with the recommendations of "Red-Book compatible audio CDs". WaveLab can import tracks directly from CDs in digital format without using sound cards ADC/DAC and sound quality damage. WaveLab software has a comprehensive set of real time effects such as Noise Shaping, Dithering, DeNoiser, DeClicker, Chorus, loudness maximizer, delay, reverbs, parameter equalizer, phase inversion. Sound processing is possible without hard disc recording in full duplex mode. WaveLab supports Audio-Access, a professional music data base that structurally organizes all your records so that you can promptly find any one kept on the CD-ROM or hard disc and play it back without boring downloading.
CoolEdit Pro is functionally similar to WaveLab 3.0. It also enables 24-bit real time data processing, all the set of sound effects, MPEG format data compression, but it is several times as cheap.
EDS TOOLS software for analyses, syntheses and processing of audio files is also worth mentioning. It is designed for well-forgotten MS-DOS and has got an excellent use's interface in Windows 95 stile. EDS TOOLS enables statistical analyses and processing of 8.. 32-bit floating-point data with 1024-channel equalizer, digital infinite and finite impulse response filters. Moreover one can easily realize any sound processing algorithms on built-in system of commands. EDS TOOLS functions of 3-dimension spectrum analyses based on FFT (Fast Furie Transform) and modern LPC methods are missing in the other software packages!
"Transistor", "tube" and "digital" sound distortions: legends and reality
Legends concerning the specific digital distortions destroying sound quality and not registered by any devices are so dubious as telepathy or a "transistor" sound. The essence of a "transistor" sound consists in different speed of decline of the amplitudes of non-lineal distortion harmonics and insignificant quantity of even harmonics. Fast exponent harmonic amplitudes decreasing for increasing frequencies is characteristic of tube amplifiers. In transistor ones harmonic amplitudes decrease slowly (according to 1/x ratio). Additionally, in tube amplifiers almost all the following higher frequency harmonics are masked with few first ones. Such psycho-acoustic phenomenon is laid into the foundation of MPEG sound compression standard. So, as I see it, a tube amplifier adds some first even and odd harmonics to a sound signal. Usually a Hi-End tube amplifier has got a non-lineal distortion co-efficient from 0.5% up to 3.0%. I should mention that the same principle of adding the first few harmonics to the original sound signal is adopted in exciters - studio sound processing devices. I would say that a tube amplifier is a kind of an exciter. That is why the tube amps with very little non-lineal distortions are not popular with audiophiles. They produce the sound very close to that of the transistor amps with minor non-lineal distortions. In transistor amps mask effect shows itself significantly less. Due to that we begin to hear "grange" and "sand" in the sound. A non-lineal distortion co-efficient must be considerably reduced to get an approximately "tube" quality sound. It is a complicated technical task. Presently, its decision is not always economically justified. For that reason, a tube amp made in Asia can be much cheaper than a North American or European Hi-End transistor amp with subjectively the same sound quality. Therefore, not all the American firms on the Hi-End market survived the crises brought about in the beginning of 1998 (see March, 98 "Class A" magazine).
Cheap AD/DACs are characterized by the absence of harmonic amplitudes decreasing for increasing frequencies. Having made estimations of the cheap sound cards (from $10 up to $ 30) I have found out that up to the sampling frequency divided by 2 all the harmonic amplitudes can be equal. In spite of a rather low harmonic co-efficient (0.02-0.04%) such converters efficiently "kill" the sound. More expensive models with harmonic amplitudes decreasing proportionally to 1/x provide a usual "transistor" sound. But recently developed 22..24-bit AD/DACs by
Analog Devices have got a really low harmonic co-efficient (up to 0.002%). They are employed in BOSS GX700, a digital sound processor generating, in the opinion of many famous musicians, more "tube" sound than that of many tube hi-fi amps.Picture 5.
Multi-Effects Unit BOSS GX700Several methods to "fight" digital distortions
Sometimes tube amps are employed to make a sound "alive" in the final record. I this case, a digital record is converted to an analog one, processed by tube equalizers (like TL Audio G400) or amps, turned back to digit and recorded to CD-RW. If such a record is played back via a transistor amp you might hear a positive result of the procedure. Otherwise, the double use of tube amps (on the stage of recording and playing back) can completely "kill" the sound. There were attempts to manufacture the digital model of a tube amp. Though, in my opinion, RedValve plug-in for WaveLab performs a sound comparable with that of an inexpensive tube amp. Generally saying, the advantage of a tube sound is especially distinguished for the middle frequencies diapason (200..8000 kHz) and dissipated for the high frequencies (8..20 kHz).
BOSS GX700 is a very interesting sound processor from the point of view of the imitation of a "live" sound with purely digital methods. It fully emulates a typical virtual audio recording studio in digit in real time. At first, an audio signal (from electric guitar and so) goes to a 20-bit high quality ADC. Then a digital signal is processed with the imitation of a tube power amp and equalizer. Moreover, you can choose them from the wide list of the devices really existing in the audio market. Then the signal proceeds through a speaker simulator (and again, there is a whole list of them) playing an important role to make a "live" sound. After that it goes to a reverb imitating the acoustic properties of the sound recording studios chambers. The dimensions of the room and the reverb decay can be chosen from the list and fixed up as desired. Flanger, chorus, phaser, harmonizer, pitch-shifter, delay effects are also available. Then the signal goes to a virtual microphone of any type accessible in the list. It is also possible to regulate the location of the microphone in the virtual studio. After a tube microphone preamplifier the signal, finally, passes to BOSS GX700 output. And moreover, it all works in real time! I have not yet come upon a software analogue of such gear.
A standard music CD comprises audio signals recorded as a sequence of 16-bit numbers with 44.1 kHz sampling frequency. Theoretically, it allows recording and consequent playing back of the audio signals with frequency components from 0 up to 22.05 kHz. In practice, most modern DACs enable to play back the frequencies up to 18-19 kHz without significant distortions. Digital and analog interpolation filters influence on the higher frequencies (about 22 kHz). They put them down on 40 dB and more. It brings about lineal, non-lineal and inter-modulation distortions. Such cut off frequency choice (about 18-19 kHz) is stipulated with economical reasons. Complexity and therefore the price of a digital interpolation filter increase when cut off frequency value approaches the half of the sampling frequency value.
Suppose that the music CD has been recorded with applying of over-sampling and a digital high quality filter with cut off frequency of about 21 kHz. Then, if your CD player or sound card have a cheap DAC using a "weak" digital filter with 18 kHz cut off frequency, you will get a worse sound for higher frequencies. But many even very cheap sound cards (opti-931, Acer S23) support 48 kHz sampling frequency. In that case, 20-21 kHz cut off frequency is turn on. Hence, one can achieve a more quality sound in the diapason of higher frequencies. At first, the digital sound track of a music CD should be imported to the WAV-file of a hard disk with
WaveLab or WinDac32 software. Then, you need to change the sampling frequency from 44,1 to 48 kHz with WaveLab, CoolEdit or EDS TOOLS procedures. They are based on software analogues of studio quality 32-bit digital filters. The result WAV-file can be played back with standard Windows 95/98/NT multimedia player or WaveLab software. Any time I have made such manipulations for different sound cards (OPTi-931, Yamaha SA700, Monster Sound 3D, Ensoniq Soundscape Ellite, Acer S23) I got significantly better sound in higher frequencies band. It would be ideal to get a special software module providing all the operation in real time without claiming a PC hard disk. But I have not yet heard of such a one.Hi-End and a digit
For a long time amid Hi-End fans and studio engineers it has been accepted to bitterly criticize 16-bit music CD sound quality. For all the faults of hi-fi amps, speakers and CD players it was digital format of audio signals to blame. But in the numerous articles appeared lately experts comparing equally expensive modern digital CD players and analogue players of "vinyl" often make choice in favor of banal 16-bit 44.1 kHz sound. So, let us leave fruitless attempts to prove the theorem by Nyquist wrong and try to see the recent development of Hi-End equipment. There is no notable progress in the scheme technique and element base of tube power amps but one exception. Ten years ago when I was keen on tube amps construction I could not even dream of an output transformer that would have passed frequency components from 10 Hz and up to 50 Hz. Now it comes true! Such a fantastic transformer "transforms" practically any tube power amp into a Hi-End gear (for example, Old Timer for $4700).
A CD player is gradually changing into something like a PC with CD-ROM and a sound card. For instance, the transport part of Audio Research CD2 CD player is based on computer CD-ROM hard ware. Crystal CS4329 micro scheme of 20-bit DAC is frequently used in the sound cards. Note that 16-bit audio signal processed with noise shaping, dithering algorithms and digital filters can be turned to a more bit signal. Besides more bits enable loudness control in digital domain without analog pre-amps. It also provides a sound quality improvement. Audio Research CD2 developers have paid much attention to a jitter problem. But initially they have created it themselves. Audio Research CD2 is connected to DAC with S/PDIF. Here it is the jitter problem to fight! And a skilled special processor does that. But it seems to me that it would be more reasonable to follow the ideology of PC structure and connect a CD transport component to DAC via IDE, SCSI, ISA, PCI or Fast Ethernet. It would totally eliminate jitter. It is an evident decision for computer hardware developers but revolution for Hi-End engineers! After all such CD players like Audio Research CD2 can boast with quite Hi-End sound quality. And all that - within the limits of 16-bit 44.1 kHz!
To draw the line I should say that your home PC can be easily turned to a full-featured professional recording studio capable of handling all aspects of the recording process, from tracking, editing and processing to mixing and mastering.
Dmitry Simanenkov
. mailto:digital_sound@bigfoot.comOther my Articles:
PC based Home Recording Studio
Creating Pro Guitar Sound on PC
3D SOUND on your PC: History, Theory, Hardware, Software
Sound Compression - MP3 and others
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