APT-X100: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING MIKE SMYTH STEPHEN SMYTH Audio Processing Technology Ltd Belfast, BT9 5AF, N. Ireland Details are presented of the apt-X100 16-bit to 4-bit digital audio compression algorithm for real-time transmission of very high quality audio at 128kbits/s. Designed to directly replace digital PCM hi-fi audio links, apt-X100 is implemented on a single masked-ROM DSP and boasts exceptional tolerance to bit errors, low coding delay, zero-overhead automatic data synchronisation and de-multiplexing, and an auxiliary data transmission facility. A complete half-duplex dual-channel encode/decode implementation within a single DSP is also reported. INTRODUCTION 1. APT-X100 Consumer expectation of broadcast audio quality The apt-X100 algorithm aims to code transparently has risen over recent years and been largely and in real-time very high quality digital audio satisfied with, for example, NICAM coding for signals at 4 bits per sample. The three key domestic television programmes and the components of the algorithm which collectively replacement of AM with stereo FM radio achieve this are sub-band coding, linear prediction transmissions. Since the launch of the Compact and adaptive quantisation. Disc digital audio has become largely synonymous with high quality audio, but its transmission and 1.1 Sub-band coding storage presents some difficulties to broadcasters due to the large and expensive bandwidth Sub-band coding splits up the audio signal into a required to provide such services. Nevertheless, number of frequency bands. By doing so the significant advances have been made in recent frequency domain redundancies within the audio years, particularly in digital signal processing signals can be exploited, allowing for a reduction hardware, enabling more complex, higher in the coded bit rate compared to PCM for a performance coders to be realised which allow for given signal fidelity. The spectral redundancies are sizeable reductions in the digital audio data rate. due to the signal energies in the various This reduction has allowed digital audio to be frequency bands being unequal at any instant in progressively introduced as a replacement for time. By altering the short-term coding resolution weak analogue links in the studio environment, in each band according to the energy of the sub-band signal, the quantisation noise can be This paper outlines the important techniques used reduced across all bands over a PCM coder for in the apt-X100 coding system, and briefly the same overall bit rate. A mechanism must also describes hardware currently available to facilitate be provided for allocating bits to each band in its widespread use. The performance of apt-X100 proportion to the signal variance within that band. is indirectly compared to transform-based coders on the basis of noise masking thresholds, and an On its own, sub-band coding incorporating PCM illustrative comparison with NICAM is also in each band, is capable of providing a gain over presented, full-band PCM for stationary 'non-flat' input signals, gain being defined as the ratio of the The apt-X100 audio coding system is currently in quantisation error variances at the same use worldwide in studio-transmitter microwave transmission rate. This gain increases with the links, DBS radio, satellite-based outside broadcast number of bands, and with the extent of the links, ISDN commentary links and additionally in 'non-flatness' of the input signal, that is with the numerous audio storage products such as ratio of the arithmetic mean to the geometric floppy-disk based digital audio recorders, mean of the sub-band variances. Figure 1 AESloth INTERNATIONAL CONFERENCE 41
Transcript
APT-X100: A LOW-DELAY, LOW BIT-RATE,SUB-BAND ADPCM AUDIO CODER
FOR BROADCASTING
MIKE SMYTHSTEPHEN SMYTH
Audio Processing Technology LtdBelfast, BT9 5AF, N. Ireland
Details are presented of the apt-X100 16-bit to 4-bit digital audio compressionalgorithm for real-time transmission of very high quality audio at 128kbits/s.Designed to directly replace digital PCM hi-fi audio links, apt-X100 isimplemented on a single masked-ROM DSP and boasts exceptional tolerance tobit errors, low coding delay, zero-overhead automatic data synchronisation andde-multiplexing, and an auxiliary data transmission facility. A completehalf-duplex dual-channel encode/decode implementation within a single DSP isalso reported.
INTRODUCTION 1. APT-X100
Consumer expectation of broadcast audio quality The apt-X100 algorithm aims to code transparentlyhas risen over recent years and been largely and in real-time very high quality digital audiosatisfied with, for example, NICAM coding for signals at 4 bits per sample. The three keydomestic television programmes and the components of the algorithm which collectivelyreplacement of AM with stereo FM radio achieve this are sub-band coding, linear predictiontransmissions. Since the launch of the Compact and adaptive quantisation.Disc digital audio has become largely synonymouswith high quality audio, but its transmission and 1.1 Sub-band codingstorage presents some difficulties to broadcastersdue to the large and expensive bandwidth Sub-band coding splits up the audio signal into arequired to provide such services. Nevertheless, number of frequency bands. By doing so thesignificant advances have been made in recent frequency domain redundancies within the audioyears, particularly in digital signal processing signals can be exploited, allowing for a reductionhardware, enabling more complex, higher in the coded bit rate compared to PCM for aperformance coders to be realised which allow for given signal fidelity. The spectral redundancies aresizeable reductions in the digital audio data rate. due to the signal energies in the variousThis reduction has allowed digital audio to be frequency bands being unequal at any instant inprogressively introduced as a replacement for time. By altering the short-term coding resolutionweak analogue links in the studio environment, in each band according to the energy of the
sub-band signal, the quantisation noise can beThis paper outlines the important techniques used reduced across all bands over a PCM coder forin the apt-X100 coding system, and briefly the same overall bit rate. A mechanism must alsodescribes hardware currently available to facilitate be provided for allocating bits to each band inits widespread use. The performance of apt-X100 proportion to the signal variance within that band.is indirectly compared to transform-based coderson the basis of noise masking thresholds, and an On its own, sub-band coding incorporating PCMillustrative comparison with NICAM is also in each band, is capable of providing a gain overpresented, full-band PCM for stationary 'non-flat' input
signals, gain being defined as the ratio of theThe apt-X100 audio coding system is currently in quantisation error variances at the sameuse worldwide in studio-transmitter microwave transmission rate. This gain increases with thelinks, DBS radio, satellite-based outside broadcast number of bands, and with the extent of the
links, ISDN commentary links and additionally in 'non-flatness' of the input signal, that is with thenumerous audio storage products such as ratio of the arithmetic mean to the geometricfloppy-disk based digital audio recorders, mean of the sub-band variances. Figure 1
AESloth INTERNATIONALCONFERENCE 41
SMYTH
Sub-band Gain (dB)5O
4O
3O
20
10
02 I I I I I I I I
I 2 4 8 16 32 64 128 256 512 1024 2048 4096
Number of Sub-bands
X Tin Whistle 0 Violin
z_ Classical Guitar X Church Choir
Figure 1 Variation of sub-band gain with number of sub-bands
illustrates the variation of sub-band gain with determining the relationships between thenumber of bands for four essentially stationary sub-band gain, the number of sub-bands and theaudio signals, response of the filter bank used to create the
sub-bands. The coding delay and computationalIn practical implementations of sub-band coders, complexity are two further problems that must betwo factors tend to limit the number of bands addressed, both of which tend to limit the numberemployed. First, the non-stationarity of audio of sub-bands used in a real-time coder.signals leads to an averaging of the energy acrossbands, reducing the arithmetic mean to geometric The apt-X100 algorithm use Quadrature Mirrormean ratios, and hence reducing the coding gain. Filters (QMFs) to divide the input data into fourSecond, the transmission of the bit-allocations for uniform frequency bands. These allow for theeach of the sub-bands becomes problematical. The exact reconstruction of the audio to within thekey issue in the analysis of a sub-band framework level of quantisation noise introduced, and alsois in determining the likely sub-band gain exhibit a constant coding delay across theirassociated with high quality audio and in frequency range. To reduce inter-band leakage,
42 AES loth INTERNATIONAL CONFERENCE
APT-XlOO: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING
which reduces sub-band gain, the number of filtertaps are weighted to the first QMF stage. Withjust four bands and a 64/32 tap split, thecomplexity and delay time of the filter ismanageable, whilst the sub-band gain is still 50G°ding Gain (dB)considerable.
1.2 Linear prediction 40 _,,___
Any reduction in potential gain from the use ofjust four sub-bands can be compensated for bylinear prediction, which aims to remove spectralredundancies which remain in the sub-bands. The 3o
total performance of the coder with the inclusionof linear prediction can easily match the sub-bandgain associated with the use of many more
sub-bands, without resorting to an increased filter i°i
complexity or coding delay.
Redundancy is removed by subtracting from theincoming sub-band signal a predicted signal,creating a difference signal which may then bequantised. If the prediction is accurate then themagnitude of the difference signal will be less , 2 4 B ts s2 64 t28 2s6 s_2 t0242o_84096than the original sub-band signal, allowing the Number of Sub-bandsquantisation resolution to be reduced. The degreeto which the sub-band signal is attenuated istermed the prediction gain and represents animprovement in the coded SNR over PCM. What
Figure 3 Composite sub-band + prediction gain versusis removed at the encoder is added back onto the theoretical sub-band gainquantised difference signal received at thedecoder.
coding delay and allows the predictor to followThe apt-X100 algorithm utilises backward adaption the short term characteristics of the signal withoutof the predictor coefficients which involves no transmitting side information.
The success of linear predictive coding is highlydependent on the predictability or correlation ofthe sub-band signals. Figure 2 shows a breakdown
P redict.!.on_Gai__n(d_B_! ......................... of the sub-band prediction gains obtainable forsome musical instruments. In each case the gain isreduced at higher bands, and for most instrumentsthe gain is apparent only in the first two bands.
The coding gain of sub-band coders with andwithout linear prediction in each band isillustrated in Figure 3. For this trombone signalthe 4-band coder with prediction is seen to givecomparable gain to a 64-band coder withoutprediction.
Adaptive quantisation exploits the relativelyslowly varying energy fluctuations in time which
audio exhibits by continually adjusting the_ quantiser step size dynamically to match the
_ __m signal level. This provides an almost constant andTin Whistle Trombone Alto Sax Banjo Cymbals Violin optimal signal to quantisation noise ratio
InputSignal throughout the quantiser operating range.
i Ba.d,(0-4 kHz) ,_ Band2 (4-Bk,z) The apt-X100 coder uses a Laplacian quantiserr_ Ba_d3 (8-t2kH_ mmBand_ (,2-t__,_) with backward adaption which extracts the step
size information from the recent history of thequantiser output. This avoids the problem of
Figure 2 Sub-band prediction gains (apt-X100)
AES 10th INTERNATIONAL CONFERENCE 43
SMYTH
The variation of quantiser performance in eachsub-band for varying input signals is shown in
Quantizer Gain (dB) Figure 4.40 .....................................
1.4 Composite algorithm (sub-band ADPCM)
The complete apt-X100 algorithm is illustrated inFigure 5. The input audio signal is filtered into
30 four frequency bands of uniform bandwidth.Incorporated within each band is abackward-adaptive predictor andbackward-adaptive Laplacian quantiser. The four
2B code words from each sub-band are multiplexed
into a single 16-bit word and transtnitted to the_ _ receiver. The decoder de-multiplexes each 16-bit
word and feeds each sub-band inverse quantiserlO _ I with its respective code. The output from each
t7 sub-band quantiser is a difference signal which
i must be added to the predicted signal generated
_ i I from previous samples. Finally the fourI reconstructed sub-band signals are inverse filtered
0 - - and output as 16-bit PCM samples in the timeAlto Sax Trombone Cymbals Tin Whistle Violin Banjo domain.
..- Ba,dI _o-4k.z_ ,_ Band2 _4-8k._ Table I shows the objective performance of theI_____ Band 3 (8-12 kHz) _ Band 4 (12-16 kHz)
composite algorithm averaged over four secondsof music for various monophonic signals. Theequivalent PCM bits shown in the last column ofTable 1 is a calculation of the number of bits that
Figure 4 Variation of quantiser performance with sub-bandswould be required for a linear PCM codec toobtain an equivalent quantisation noise in each
estimation delay and range transmission sub-band.over-heads, but has the disadvantage in that theadaption is based on reconstructed values and not 1.5 Coding delay of composite algorithmthe original values_ This tends to limit the
accuracy of the estimation and for non-stationary In apt-X100 the loop-back coding delay, that is theinputs, such as fast signal transients, an accurate delay between the input of an audio signal andestimation of the impending input signal is not the output of the reconstructed signal, is 122 PCMpossible, samples, and is due to the use of linear phase,
Input Band Prediction Quantisation Band Gain Equiv.Gain (dB) Gain (dB) (dB) PCM bits
APT-X100: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING
audio audio
in out
Fixed bit allocation Fixed bitallocation
% V f % v l
encoder decoder '
Q backward adaptive quantizer
1lQ inverse backward adaptive quantizer
A quantizer step-size adaptor
P all-pole backward adaptive predictor
Figure 5 apt-X100 music coder
digital filter stages. Corresponding time delays for quantisation tend to reduce the significance of thestandard PCM sampling frequencies are given in errors by spreading their affect over the trailingTable II. window of samples used for the updating.
Furthermore, the magnitude of the effect of a bitFor audio storage/playback and general error on the sub-band predictors and quantisers isbroadcasting, delay times are not critical, proportional to the magnitude of the differentialHowever, for applications such as audio signal being decoded at that instant. Thus if theconferencing over ISDN, or outside broadcasting transmitted differential signal is small, which willwhere the functionality of the service is greatly be the case for a low level input signal or for aenhanced if delay times are kept short, apt-X100 is resonant, highly predictable input signal, then anyideal, bit error will have a small effect on either the
predictor or quantiser at the decoder. The bit error1.6 Bit error response response is thus well-matched to the sensitivity of
human hearing, traditionally critical signals beingThe apt-X100 algorithm is inherently insensitive to relatively immune to errors.random bit errors, and therefore no protection ofthe transmitted compressed data is necessary. No The propagation of any error is halted at theaudible distortion is apparent for normal decoder through the use of exponentially-taperedprogramme material at a bit error rate (BER) of windows which give greater importance to the1:10,000 while speech is intelligible at a BER of most recent samples and allow the influence of1:10. samples further back in time to 'leak' away.
This resilience to bit errors is due to a 1.7 Objective comparison with NICAMcombination of the three coding elementsdiscussed. First of all distortions introduced by bit The performance of the BBC's NICAM codingerrors are constrained within a sub-band. In system is well documented and provides a usefuladdition backward adaptive prediction and point of reference for more recent audio coding
AES 10th INTERNATIONAL CONFERENCE 45
SMYTH
PCM Sampling Time Delay Between Input AudioFrequency and Output Reconstructed Audio
Figure 6 apt-X100 sub-band coding resolution compared to NICAM resolution
46 AES loth INTERNATIONAL CONFERENCE
APT-X100: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING
systems [l]. Figure 6 compares the objective the original signal that falls below this maskingperformance of the individual lower and upper threshold may be ignored, and in addition thosesub-bands of apt-X100 with NICAM for a parts that remain can be quantised to the level oftrombone input signal, the energy of which is also noise indicated by the threshold at that particulardisplayed. The objective performance is again frequency. Thus the noise masking thresholddisplayed as the 'equivalent PCM bits' that would serves to remove irrelevant spectral parts of thebe required to produce the same coding noise signal and to reduce the level of coding accuracylevel. In the low energy regions of the passage the needed for the remainder.equivalent PCM resolution of the lower (0-4kHz)band is very similar to NICAM, with apt-X100 ADPCM on the other hand is a procedure whichdropping slightly below NICAM during the aims primarily to determine and removelouder sections. These results imply that the 16-bit redundancy from an audio signal, that is, toto 4-bit coding performance of apt-X100 remove those predictable parts of the signal priorapproaches that of the 14-bit to 10-bit coding to coding which may be reconstituted again, so asscheme of NICAM up to 4kHz. However, the to accurately reconstruct the original signal [3]. Asobjective performance of apt-X100 equals or such ADPCM does not explicitly take into accountsurpasses that of NICAM above 4kHz for this the auditory properties of the human ear,passage. This is clearly seen in the equivalent redundancy being a purely objective measurementPCM bits measured in the upper bands, an of any signal. Quantisation noise introduced to theexample of which is illustrated in Figure 6. This signal during coding is added without anyshows the objective performance of the 12-16kHz knowledge of the allowable noise maskingband which exceeds NICAM by almost 2 bits thresholds. The apt-X100 algorithm, by using fourduring the loudest sections. The problem of frequency bands, simply constrains quantisationaudible noise pumping at high frequencies, to noise to within these bands. It is of interestwhich broad-band coders such as NICAM are therefore to compare the noise injected bysusceptible, is thus eliminated in the apt-X100 apt-X100 with the noise masking thresholdssystem. Hence the subjective performance of calculated from a psycho-acoustical model ofapt-X100 should exceed that of NICAM for this hearing.material.
The model used to calculate the noise maskingthresholds is based on Johnston [4] and involves
2. MASKING THEORY APPLIED TO transforming the signal into the frequency domain,APT-X100 partitioning the power spectrum into 'critical
bands', calculating masking thresholds in eachThe outstanding audio performance of the band and finally spreading the thresholds of eachADPCM based apt-X100 algorithm is not band across all bands. This procedure forsurprising given the development of ADPCM over determining the noise masking thresholds doesa long period of time as the standardised method not take into account temporal masking effects.of speech compression [2]. ADPCM consists ofcoding elements which are inherently matched to Figure 7 is a spectrum of the glockenspiel (EBUthe auditory process, and the inclusion of ADPCM SQUAM CD 422-204-2 track 35) and its associatedwithin a sub-band framework further improves noise masking threshold, and Figure 8 illustratesthis. Optimisation of the performance of the the error spectrum generated from apt-X100 forapt-X100 algorithm has relied on extensive this signal. The frequency scale is anlistening tests, without any direct reference to approximation to the bark scale, each critical bandauditory theory. In this section the coding being of uniform width. The most importantperformance of apt-X100 is compared to noise feature is that the error signal is generally belowmasking thresholds, the calculation and application the masking threshold at all frequencies, and isof which form the basis of many transform-based typically flat across each of the four sub-bands. Ataudio coders, the resonant frequencies the noise is considerably
below the allowable threshold. This is intuitivelyIn digital audio coding an auditory masking reasonable given that resonant signals tend to betheory is usually invoked in an attempt to predictable and hence exhibit redundancy whichdetermine the irrelevant parts of an audio signal, may be removed from the signal. What is perhapsIrrelevant parts of the signal are those which are surprising is that the error signal is below thedeemed inaudible to the human ear, due to their masking threshold even in the noisy regions ofbeing masked by higher level signals at different the spectrum, where the predictability andand usually lower frequencies. To determine redundancy of the signal is small. It is alsoirrelevancy involves characterising and modelling noteworthy that while the use of the maskinghuman hearing, and applying this model, or an threshold will tend to remove large portions ofapproximation of it, to the audio signal. The final the spectrum, particularly at the high frequencies,result is a frequency-dependent masking threshold, apt-X100 will always code these signalwhich gives an indication of the level of noise at components, irrespective of their level.any frequency that may be injected into theoriginal signal without audible effect. Any part of In this particular spectrum therefore, apt-X100,
AES 10th INTERNATIONAL CONFERENCE 47
SMYTH
Frequency (kHz)
Energy (dB) I 2 3 4 5 6 7 8 12 15
9o I I I I I I I I [ [ I III
8O
7O
60
50--
40 ---
-20 __ _t
10
0
Noise ,masking threshold for each Bark band '-Figure7 Spectrum and masking thresholds of glockenspiel
Frequency (kHz)
Energy (dB) I 2 3 4 5 6 7 8 12 15
I [ [ I I [ [ I [ I [ III[
90--
80--
70--
60--
50--
40----
30--
,Oo v,..,._ t_i_ ,_r_,,_l_,t_,'_tl t _, 'Noise masking threshold for each Bark band --'
Figure 8 apt-XlO0 error spectrum of glockenspiel
48 AES loth INTERNATIONAL CONFERENCE
APT-XlO0: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING
Frequency (kHz)
Energy (dB) I 2 3 4 5 6 7 8 12 15I I I I I I I I I I I IIII
B
90 l
6O
5O
40
30-- ]_
20--
10--
0 --
Noise masking threshold for each Bark band "-Figure9 Spectrum and, masking thresholds of trombone
Frequency (kHz)
Energy (dB) I 2 3 4 5 6 7 8 12 15
I I I I I I I I I I I IIIII
90--
80--
70--
60----
50--
40--
30--
',,,· ,. t _l II, 1
0I
Noise masking threshold for each Bark band mFigure 10 apt-XlO0 error spectrum of trombone
AESlothINTERNATIONALCONFERENCE 49
SMYTH
Frequency (kHz)
Energy (dB) 1 2 3 4 {5 6 7 8 12 15
_ I I I [ I I I I I I IIIIII I
I9O
80
10
0I I I I I I
Noise masking threshold for each Bark band '-
Figure11 Spectrum of glockenspiel (transient)
FreqUency (kHz)
Energy (dB) I 2 3 4 5 6 7 8 12 15
I I I I I I I I I I I IIII
0' ""
80--
70--
60--
50--
i '""lI't2O
10
0-I I I i I I I
Noise masking threshold for each Bark band '-Figure12 apt-X]O0 error spectrum of glockenspiel (transient)
50 AES loth INTERNATIONAL CONFERENCE
APT-XIO0: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING
Frequency (kHz)
Energy (dB) I 2 3 4 5 6 7 8 12 15
I I I I I I I I I I I III1
90--
80--
70--
60--
50--
,o_30-- I __
0 '} Il'
Noise masking threshold for each Bark band --
Figure13 apt-XlO0 error spectrum of glockenspiel (5 passes)
whilst not fully exploiting the noise masking the masking threshold and become audible.threshold in the resonant regions, nevertheless Coders which rely on a calculation of the noisedoes not exceed it elsewhere. This is also seen in masking threshold for their operation counter thisFigures 9 and 10 which show the spectrum, error problem by a more conservative use of thespectrum and noise thresholds of a trombone masking theory, for example, by reducing the(EBU SQUAM CD 422-204-2 track 22). level of noise introduced into the critical bands
containing resonant structure [5]. This allows for aFor a transient signal (Figure 11), also of a limited degree of tandem coding albeit with aglockenspiel, the error signal (Figure 12) is slight audible degradation between passes and atconsiderably increased and tends to rise above the the expense of less compression. The apt-X100noise threshold typically in the 8-12kHz sub-band, algorithm, by not fully exploiting the noiseThis general increase in noise is also reasonable masking threshold especially in resonant regions,given the unpredictability of transient signals, is much more tolerant of tandem coding. This isHowever, the audible effect of this noise demonstrated in Figure 13 which shows the errorbreakthrough is wholly alleviated by temporal spectrum of a glockenspiel signal (Figure 7) aftermasking, a psycho-acoustic phenomena which is five digital passes through the apt-X100 algorithm.explicitly accounted for within the apt-X100 Each apt-X100 coding passage accumulates noisealgorithm by the fast-attack/slow-decay but at a decreasing rate leading to a gradualcharacteristic of the adaptive step-size of the degradation in audio quality with the number ofquantiser, codingpasses(Figure14).
Given that these examples show that apt-X100 In general apt-X100 is well-matched to the overalldoes take account of the characteristics of human characteristics of human hearing in that criticallyhearing without any explicit calculation of the perceived signals such as pure tones are highlynoise masking threshold, what advantage is redundant and hence accurately coded, whilstgained from not fully exploiting the threshold? noisy signals which are not predictable andSome applications require an ability to therefore poorly coded are not critically perceived.post-process the compressed data in the digital Comparison with noise thresholds calculated fromPCM domain, and subsequently re-code the data. a masking theory show that apt-X100 inducedIf all irrelevancy is removed from an audio signal noise is consistently below audible levels.in the first coding pass, then by definition tandem Moreover the audio bandwidth of apt-X100 iscoding will cause the cumulative noise to exceed simply one half of the sampling rate, irrespective
AES loth INTERNATIONAL CONFERENCE 51
SMYTH
Apart from the main sub-band ADPCMSNR (dB) processing blocks, all apt-X100 devices
50 also include many additional softwarefeatures to enhance their functionality.
These include automatic synchronisation,auxiliary data transmission and PCM
_ ----_ leveldetection.
....... Automatic synchronisation is a facility- O_J
which enables the apt-X100 compresseddata stream to be decoded without priorknowledge of the 16-bit compressed wordboundaries. As a result the use of
............ 20 AutoSync enables the compressed data tobe handled at both transmitter and
receiver using only bit timing, no wordclock being required. No bandwidthoverheads are incurred through the use
10 of any of the AutoSync facilities.
Synchronism is obtained by inserting intothe compressed audio data stream aunique sync word. This is searched for at
i , , i i , , 0 the decoder and once found establishes2 3 4 5 6 7 8 9 10 the compressed word boundaries for
Coding passes certain multiplexed formats. Insertionoccurs once every 128 16-bit compressed
Figure 14 Variation of SNR with coding passes (apt-X100) words (frame length = 2048 compressedbits), and to establish synchronisation
of its absolute value. This is of relevance in three consecutive sync words must be found byapplications which demand operation at different the decoder. This allows for synchronisation undersampling rates, adverse channel error conditions as summarised in
Table III.
3. HARDWARE IMPLEMENTATION Loss of sync lock is flagged at the decoder, andthe PCM output muted if sync words are not
3.1 Chip level found in three consecutive frames. Synchronisationfrom processor power-up or bit errors is therefore
The apt-X100 coding system is currently available achieved within two to three frame periods, whileon a single AT&T DSP16 digital signal processor re-synchronisation is achieved within four to sixwith separate devices being required for the frames from the instant of clock slip. The timeencoding and decoding stages, the APTX100E and taken to (re)synchronise is thus entirely dependentAPTX100D respectively. With 512 X 16 words on the sampling rate.RAM and 2048 X 16 words of program ROMincluded in the DSP16, it has been possible to In addition to synchronising a single channel,store and run either encoder or decoder AutoSync also enables two, four or eight channelsalgorithms on masked ROM versions of this to be multiplexed/de-multiplexed, again with nodevice without the need for external memorv. As bandwidth overheads. Only one encoder processora result, the hardware design around the ap(-X100 is used to send a sync word, whilst each decoderprocessors is straightforward, requiring normally processor uses this sync word to de-multiplex anyonly the PCM convertor and some basic timing one channel out of the original two, four or eightlogic to complete an audio data compression channels.system.
AutoSync combines powerfully with the 16-bitRecently dual channel apt-X100 encoding and word format of the compressed audio data stream,decoding algorithms have been implemented on a which is essentially identical to standard 16-bitsingle ROM-masked AT&T DSP16A digital signal audio PCM. For example, the AutoSync facilityprocessing chip. In stereo applications, the use of enables the direct replacement of a single 16-bit,a single DSP16A device instead of two 32kHz sampling rate, 1024kbit/s stereo audio linkmonophonic DSP16 chips, is particularly economic with four apt-X100 compressed stereo channelsin high volume projects, with no change in timing circuitry.
52 AES 10th INTERNATIONAL CONFERENCE
APT-XlO0: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING
Table III AutoSync lock failure under random bit errors
3.3 Auxiliary data frequency of the ADC may be set either internallyby an on-board oscillator, (active clock operation)
Auxiliary data may be transmitted between the or externally by an input clock, (passive clockencoder and decoder chips by replacement of the operation). Auxiliary data is supported as TTLleastsignificant bit of each 16-bit compressed data input from a synchronous serial port.words. This bit stealing can be dynamicallyswitched on or off according to line activity The stereo decoder board operates passively only,through the use of the AutoSync which has the the minimum requirement for operation being bitfacility of allowing the encoder to remotely enable clock timing, to establish the sampling rate, plusthe auxiliary data mode of the decoder. The data compressed data (AutoSync enabled). Thebit rate for transmission of auxiliary data per compressed input data stream may comprise one,mono channel is 1/4 of the sampling rate, two, four or eight multiplexed channels, any twoallowing standard 9600 baud transmission within of which can be selected for decoding. Audio isa 32kHz stereo audio link, and up to 24kbit/s output either in analogue from the dual-channelwithin a 48kHz stereo link. This opens up the 16-bit DAC, or in digital format (AES/EBU orpossibility of a talk-back facility during normal S/PDIF), or from a remote DAC (direct TTLaudio transmission between transceivers, toll output of PCM). All timing and compressed dataquality speech (3400Hz bandwidth) being signals are RS422 (balanced). With AutoSynccommonly realised at 16kbit/s. disabled, a word clock must be supplied to define
the compressed code word boundaries. Auxiliary3.4 PCM threshold level detection data is output as TTL.
This facility compares the absolute value of the 3.6 PC expansion card (ACE100)input PCM signal averaged over four sampleswith one of 64 possible threshold levels held in a The ACE100 PC expansion cards have beentable within the encoding processor. If the PCM recently introduced and are primarily intended aslevel is above the threshold level a THRES flag is a low cost platform on which OEM's can developset high. Allowed threshold levels range from their own application specific software. The card-96dB in 1.5dB steps and are set by applying the essentially allows two channel record andappropriate table address to the encoder device, play-back of apt-X100 compressed audio or 16-bit
PCM audio at user-selected sampling frequencies3.5 Stereo coding/decoding boards directly to and from a PC hard or floppy drive. A
noteworthy feature of the ACE100 is the inclusionStereo encoder (SCS100) and decoder (SDS100) of a user-programmable DSP which has directboards have been developed which allow easy access to all the main data busses. As a result,access to every facility offered in the apt-X100 both pre- and post-processing of the audio data isalgorithm, and are primarily intended for OEM's. facilitated. Up to four cards may be usedUse of 64 times oversampling dual-channel A/D simultaneously in any single PC, allowing theand D/A convertors allow sampling frequencies to possibility of an 8-track digital audio recorder.vary continuously between 8kHz and 48kHz withno additional filtering. The boards are designed to 3.7 Duplex stereo transceiver (DSM100)operate in either stereo or mono.
For broadcasters a complete stereoThe stereo encoder board accepts either balanced encoding/decoding unit has been developed, theanalogue audio or digital audio (AES/EBU or DSM100. This unit utilises the stereo codec boardsS/PDIF formats) inputs, with a further option of described above. Audio I/O is either balancedreceiving direct TTL data from a remote ADC. analogue or digital (AES/EBU or S/PDIF), withThe compressed output data is in 16-bit word an unbalanced microphone input and headphoneformat, each word representing four PCM output also provided. Compressed data I/O is viasamples. Data output and all output timing a RS449 interface, the data rate being determinedsignals are RS422 (balanced). The sampling externally by the clock frequency of the
AES loth INTERNATIONAL CONFERENCE 53
SMYTH
Processor VersionMode (AT&T DSP16A cycle time)
(encode or decode) 55 nsec 33 nsec 25 nesI
32 kHz mono yes yes yes44.1 kHz mono no yes yes48 kHz mono no yes yes
32 kHz stereo no tba(*) yes44.1 kHz stereo no no tba(*)48 kHz stereo no no tba(*)
Table IV Modes of stereo encode/decode processor
transmission line, or internally by an on-board applications which normally transmit or storecrystal oscillator clock on the encoder card. A full PCM digital audio signals.duplex RS232 auxiliary data interface is alsoprovided, allowing up to 19200 baud transmission The apt-X100 process is based on sub-bandbetween DSM100 transceivers (sampling frequency ADPCM techniques which utilise both signal48kHz). Visual indicators of the input and output redundancy and auditory masking to achievestereo signal levels, current mode of operation, transparent coding at 4-bits per sample. This hasstatus and errors are provided, relevant signals been demonstrated by comparing coding noisebeing fed to an external alarm port. introduced by apt-X100 with noise masking
thresholds calculated from a relevant auditorymodel. Moreover, because the noise levels are
4. STEREO ALGORITHM generally well below the calculated values,apt-X100 has been shown to be tolerant of tandem
A very recent development is the implementation coding.of a stereo encoder and stereo decoder algorithmon a single masked version of the AT&T DSP16A A summary of the operational features of thesignal processor. This processor has 2k x 16 RAM apt-X100 system must include:and 12k X 16 ROM which is sufficient internal
memory for storing two monophonic encoder · subjectively and objectively optimisedalgorithms and two monophonic decoderalgorithms, and for running the chip either as an ° operation at any sampling frequency toencoder or decoder in stereo or mono. The range 48kHzof operating modes of this signal processor are
outlined in Table IV. ° inherent tolerance of tandem coding andpost-processing
Furthermore a more efficient implementation ofthe algorithm which is under development will · inherent immunity to bit errors better thanenable the 33ns version to encode or decode 1:10,00032kHz stereo and the 25ns version to process 44.1and 48kHz stereo audio. ° very low coding delay (< 4ms at fs = 32kHz)
The stereo encode/decode chip can dynamically ° linear phase response and low ripple acrossswitch its mode of operation from encoding to bandpassdecoding. In applications such as audio
storage/playback this ability to swap operating ° low hardware complexity and costmodes makes the chip very cost effective.
apt-X100 is currently available in four formats toallow entry of the technology at any design level.
5. CONCLUSION
1. The lowest level is pre-programmedapt-X100 is a very high quality 16 to 4-bit audio encoding and decoding chips (Figure 15),compression system designed specifically to which allow OEMs to implement and specifyreduce by a factor of four the amount of data apt-X100 without the need to formally licenseassociated with PCM digital audio without the technology. Each chip incorporatesaffecting the sound quality. Use of the system features such as automatic dataallows an enormous saving to be made in those synchronisation and de-multiplexing schemes
54 AES loth INTERNATIONAL CONFERENCE
APT-X100: A LOW-DELAY, LOW BIT-RATE, SUB-BAND ADPCM AUDIO CODER FOR BROADCASTING
to facilitate zero overhead decoding of data post-processing is supported through the usein asynchronous environments. Hence for of a user-programmable DSP which controlsmany applications only limited additional all data movements between the ADC, DAC,timing circuitry is necessary apt-X100 devices and main memory
2. The next level is based on stereo encoding 4. The highest level is a fully integrated digitaland decoding circuit boards (Figure 16) audio transceiver unit (Figure 18). Inwhich are completely self-sufficient, having addition to the facilities offered by the stereoon-board audio conversion, clock encode/decode boards this unit offers RS449regeneration and data I/O circuitry. Both compressed data I/O, RS232 auxiliary data,boards have been designed with OEM's in and a choice of balanced analogue or digitalmind being capable of operating in a broad (AES/EBU S/PDIF) I/Orange of applications
apt-X100 is today a widely used audio coding3. For ease of use within the PC environment system, being specified in many products in the
a PC-based expansion card (ACE100 Figure broadcasting and professional audio markets. This17) has been designed which allows for the proliferation is primarily due to its singular highreal-time recording and play-back of quality, wide applicability and Iow hardwarecompressed stereo audio directly onto complexity.either floppy or hard drives. Pre- and
B
[]
Figure 16 Stereo encoder and decoder cards Figure 18 DSM100 digital audio transceiver
AESlothINTERNATIONALCONFERENCE 55
SMYTH
REFERENCES
[1] Soumange J., Mabilleall P.: "A comparative [4] Johnston J.D.: "Estimation of perceptualstudy of the proposed high quality coding entropy using noise masking criteria"schemes for digital music" Proc. ICASSP, ICASSP-88 New York pp. 2524-2527, AprilApril1986 1988
[2] CCITT, Study Group XVIII, Report R26(c)n [5] Thiele G., Stoll G., Link M.: "Low bit-rateRecommendation G.72x, August 1986 coding of high quality audio signals. An
introduction to MASCAM"
[3] Smyth S.M.F.: "Digital audio compression - EBU Review - Technical No.230, Aug 1988a practical solution" Proc. NAB EngineeringConference 1989, pp 234-241
