Audio Engineering Society UK

Thomas Lund, TC Electronic A/S

Thomas Lund’s background includes work as a recording engineer and musician and the study of medicine – an unusual combination which may contribute to his understanding of loudness perception. Thomas has also been involved the design of many TC Electronic’s products, he has contributed to various standardisation groups on the subject of loudness, and has authored many papers presented to the AES and other bodies.

Traditional Loudness Measurement

Recent years have seen the ‘level’ of pop/rock music, as delivered by CD, steadily increase. Thomas cited the simple way that audio level has been measured as a partial cause. Historically, audio level has often been measured by peak programme meters, and commonly used definitions of overload have been very simplistic methods such as peak-level-counting (eg three consecutive full-scale samples equals overload). Such simple techniques of measuring (and by association, limiting) the level may have worked well when systems consisted of a microphone, a preamp and an ADC but with digital processing techniques numerous methods have been devised to increase the apparent loudness of material delivered on CD while ‘working around’ the peak-level limitations, apparently (we must assume) to some perceived commercial benefit to the record industry.

Many hold the opinion that such ‘hot mastering’ techniques are severely detrimental to the overall quality of modern music releases. Thomas calls this drive for increased level whatever the cost, coupled with a high willingness of broadcasters and consumers to use large amounts of data compression (for archiving, broadcast and replay), a ‘war on music’.

The Problems of Incorrect Levels

With such hot-mastering techniques, it is trivial to generate digital signals that exceed 0dBFS in the analogue output, after the assumed reconstruction or up-sampling filters. The greater-than-0dB peak levels can cause serious problems in the reproduction chain where some processes have been implemented with the assumption that 0dBFS is the largest signal they should expect.

Thomas offered demonstrations based on a commercially available ‘professional-grade’ sample rate converter, subtracting output from input. In this experiment the output should have been silent but differences could be heard clearly, manifested as ticks and signal-related noise. Other potential problem areas, according to Thomas, include limiting in mix-busses and codecs such as MPEG 1 layer 3 . These processes can all exhibit similar problems when faced with very high level inputs, a phenomenon Thomas further demonstrated. The codec problems can depend on the implementation of the codec as well as the codec itself.

Because of these issues, Thomas recommends normalising to -3dBFS – not to 0dBFS, in digital mixing and recording situations. He pointed out that the final 3dB increase can be done in the mastering room without any real quality loss, given that most recordings use 24 bits.

Better Methods of Loudness/Level Measurement

Thomas gave a functional summary of various improved methods of measuring loudness level and showed relative results based on ITU-R BS.1770. A simple improvement is the over-sampling peak programme meter which offers a more accurate representation of the true peak level.

Thomas also presented a loudness meter available from TC Electronic as a plugin for Pro Tools as ‘LM5 Loudness Radar Meter’.

TC LM5 Loudess Radar Meter

This meter includes representations described as ‘Loudness Units’ (LU) or LkFS, ‘Consistency’ and ‘Center of Gravity’, where Center of Gravity indicates the overall loudness of the programme material or music track, and Consistency indicates the ‘intrinsic loudness changes’ present in the track, with 0 representing a steady-state signal (one which has no loudness changes at all, eg a sine-wave) and progressively more negative numbers indicate reducing Consistency. Low Consistency scores such as -4 or lower indicate that the material may have a large dynamic range.

Conclusions

In conclusion, Thomas offered the following recommendations:

• Stop Counting Samples: There are better methods of measuring peak levels than counting the number of consecutive full-scale samples
• True Peak Level: Set maximum peak level at -1dBFS using a true peak meter equipped with oversampling capability.
• Dialog Level: Suggested level of dialog is -26 to -22 LkFS.
• Music: Suggested level of music is -20 to -20 LkFS.
• Avoid Peak level normalisation

If audio level is anchored only to peak level or only to dialogue, both commonly used techniques, loudness chaos is likely to ensue with extreme level jumps between programme, commercials and other home sources.

The tools and understanding exist to provide well-balanced loudness levels between different programmes and material, providing the end-listener a more pleasant viewing/listening experience and the potential for reduced distortion and overall quality improvement. Thomas outlined the problems and offered tools and methods for solving them.

Report by Nathan Bentall (edited by Keith Howard)

Title: ‘Reality is Not a Recording / A Recording is Not Reality’
Location: Royal Academy of Engineering
Description: Jim Anderson of Jim Anderson Sounds
Start Time: 19:00 for 19:00
Date: May 12th 2009

Abstract

The former New York Times film critic Vincent Canby wrote: “all of us have different thresholds at which we suspend disbelief, and then gladly follow fictions to conclusions that we find logical.” Any recording is a ‘fiction’, a falsity, even in its most pure form. It is the responsibility, if not the duty, of the recording engineer, and producer, to create a universe so compelling and transparent that the listener isn’t aware of any manipulation. Using basic recording techniques, and standard manipulation of audio, a recording is made, giving the listener an experience that is not merely logical but better than reality. How does this occur? What techniques can be applied? How does an engineer create a convincing loudspeaker illusion that a listener will perceive as a plausible reality?

Meeting Report

Jim Anderson: Professor of Recorded Music, Clive Davis Department of Recorded Music, New York University

Jim started his lecture with the attention-grabbing statement that audio recording is trickery, a devious deception – then expanded the point to explain that the aim is to make you, the listener, believe you’re hearing the truth: but actually it’s sleight of hand. He set about illustrating that by playing back a diverse range of audio recordings over the course of the lecture and discussing them, casting some light onto the techniques and tricks he’d used to exercise that devious deception: and without exception, create musical listening experiences of quite exceptional quality.

Jim started by playing the commercial release of J. J. Johnson’s “The Brass Orchestra” – it was extremely punchy, dynamic, and live-sounding. He then played another track: while obviously the same piece, and possibly the same very performance, it had much less impact, drums were much quieter, the soloist was clearly off-mic – this was from a simple stereo pair of mics to capture the “air” of the room, and illustrated the striking difference between the somewhat artificial, yet highly-appealing experience created by the commercial release, and the fly-on-the wall experience of the performance – which is arguably the “real” experience. Jim then discussed some of the details of this performance and the techniques he’d used to create the “false”, yet plausible and appealing final product: it was captured live-performance-style in a single take with no overdubbing; microphone selection was key in realising tonal and dynamic differences within the group; the studio had a “good” acoustic for performance, but this was enhanced with artificial concert-hall reverb. The artist wanted to mix first without the solos, in order to get all the internal balances right: then add the solos later – so the whole thing was mixed twice.

Jim expanded on the microphone selection points by playing “High Noon – The Jazz Soul of Frankie Laine” featuring Gary Smulyan, baritone sax player. Jim used ribbon microphones, with their smooth, easy sound, on all the nine-piece backing group; but used condensers to bring the baritone sax and French horn into sharp dynamic focus. It allows the backing to be up-front in the mix, yet keeping the sax solo sounding appropriately prominent.

To illustrate another interesting technique, Jime played drummer Marvin “Smitty” Smith tracking “The Road Less Travelled”: Marvin had requested “more depth, more breadth” in the kick drum. Jim met this requirement by using a Beyer Opus 51, a boundary effect mic designed for piano, under a sheet of wood to isolate it from the rest of the kit. He used two Opus 51s and an M88 in the middle, to create a mid/side array. In stereo, it creates perfect image of the kit: in mono, it collapses and provides a remarkably leakage-free kick drum.

Among other recordings Jim discussed, he played a track by Patricia Barber, recorded in Chicago. It had an extraordinarily huge, deep, broad-sounding kick drum, very prominent and snappy drums in general, whereas the female vocal is up-front yet full in the low-mids. He then played another recording, with same trumpeter in the same room, yet smoother-sounding – because it’s a tube mic rather than ribbon. Kick drum is only 18”, but with good tuning and an M/S mic it gives the huge depth and finish.

All recordings played so far had been tracked straight to digital: Jim’s next recording was a modern attempt to recreate the classic 1970s Blue Note sound, for an album called “Hubsound – The Music of Freddie Hubbard” Contrary to direct-to-digital tracking, this was done using a 16-track 2” at 15 inches-per-second with no noise reduction. It’s impossible to make lots of overdubs because 16 tracks is very limited. In this way, it emulates not only the sound, but also the practical constraints and therefore the recording techniques, of the Blue Note vintage.

Next up, we heard Gonzalo Rubelcaba performing “Here’s that Rainy Day” in Criteria Studio A in LA: solo piano in a large live rectangular room. Mics were a U87 above, DPA4007 close, DPA 4006 a little further back: and beyond that, a pair of U87s in a modified polyhymnia configuration, so the room sound was also captured in case a surround mix was subsequently needed.

He then played for us Bebo Valdes, a live recording done in a recording truck at the Village Vanguard nightclub. Mics were just a Sanken CUW180 with pair of ratchet movable capsules, here set up for X/Y. Mic pres with A-D were on stage, plus an audience microphone, and optical links connected the A/Ds to the truck. The recording setup was triple-redundant with Tascam DA98s, but the primary recorder was ProTools HD. Jim created a rough mix on Yamaha DM2000, for the performers to check each performance immediately afterwards. Mics were a combination of omnis and cardioids on piano, the Sanken X/Y on bass, and omnis on audience. The worth of the latter was shown when the audience start singing along – precise capture of the audience really added atmosphere to the final product.

Jim concluded by playing us his first ever jazz recording – Ella Fitzgerald at the New Orleans Jazz and Heritage festival 1977, knew Stevie Wonder was in the audience, so called him up to join in! The encore was the duet “You Are The Sunshine Of My Life”. It was a pretty magical moment to capture for a first jazz recording: particularly as immediately after the end of the song, the tape ran out, right then! A close-run thing.

Jim wrapped up this interesting talk – and listening session – by maintaining he’s the liar! Thanks to PMC and Arcam for the superlative audio reproduction system kindly lent to us for the evening.

Meeting report by Michael Page

Gert Munch, Bang & Olufsen

During this lecture Gert Munch will demonstrate how the development of several key technologies, including the development of “acoustical lenses,” led to the design and implementation of the BeoLab 5 loudspeakers.

Gert is based at the Acoustics Research division of Bang & Olufsen, Denmark; he is a specialist in electro-acoustics and has worked at B&O for 30 years. In that time he contributed to the development and design of numerous speaker models, including the subject of this evening’s lecture, the BeoLab 5.

The aims for the BeoLab 5 design included

• to make the best possible loudspeakers with the most convincing total sound experience
• to give best possible experience wherever you sit, wherever the loudspeakers are placed
• to reproduce the whole audible spectrum and dynamic range
• to make a loudspeaker that didn’t sound like one!

In order to realise the ambition, the following requirements were specified:

• Adaptive bass control including a moving microphone measurement system
• Active loudspeaker design using high power ICE power amps
• Thermal compression compensation (to remove temperature dependency of response)
• Advanced thermal protection including thermal modelling and monitoring
• Precise mechanical control and fitting for consistency
• DSP Processing for response correction and manufacturing variation control

A little history: In the mid-1980s, B&O made the ‘Penta loudspeaker, which embodied the early attempts at B&O to take control of speaker directivity. It had a tapered design, with centralised tweeters, to minimise effects of the floor and ceiling reflections, a factor recognised by B&O engineers as critical to the sound in a real room.

To further understand these reflection issues, the Archimedes project was established (running from 1988 to 1992), and carried out by B&O in conjunction with the Technical University of Denmark and KEF (the UK-based loudspeaker manufacturer). This work led to many ideas about improving loudspeakers and a new, improved unit was designed that, unfortunately, never made it to market.

BeoLab 5 evolution: Also around this time, Sausalito Audio Works was pioneering loudspeaker design incorporating what it dubbed ‘Acoustic Lens Technology’. Despite some initial scepticism, B&O engineers concluded that the speakers from Sausalito actually sounded good.

After several iterations at B&O of the initial Sausalito design, the BeoLab 5 was the evolutionary result. Its distinctive shape (some liken it to a Dalek or a pylon) make it easily recognisable – and it weighs in at a hefty 61kg!

The Acoustic Lens (perhaps a ‘lens’ in the sense that a curved-mirror in a reflector telescope can be a lens) is a mechanical structure that consists of a specially shaped reflector mounted atop an upward facing driver, the special shape being a quarter of an ellipsoid.

An ellipse has two focal points; the drive unit is located at the first so that, by virtue of the shape, all sound passes through the second (assuming a ray-tracing model and an infinitesimal source).

Prior to the building of the speaker, some ray-tracing based simulations were attempted. This simulation technique was later abandoned because such a basic model lacks the ability to predict diffraction effects, a critical factor in loudspeaker directivity.

An audience member asks, ‘Why not place the speaker at the second focal point and do-away with the lens?’. Gert’s answer is that such an approach would not provide any control over the radiation pattern – and it is this radiation pattern control that the ‘acoustic lens’ technology seeks to master.

Later modelling attempts included Boundary Element and Finite Element Analysis.

An animated picture is shown to demonstrate a radiation pattern simulation. The key point is that the response looks the same at a wide range of angles in the horizontal plane. Comparing the two-dimensional Finite element model with the 3-dimensional boundary element model, it is noted that, as presented, they look very similar, providing further confidence in their validity and the concept in general.

Gert points out that, at least initially, the ideal radiation pattern of this speaker appears to be similar to that of a dipole, however, the problem of traditional dipoles loudspeakers is they must be placed at least 1m away from the wall behind to achieve good performance, a restriction which can prove inconvenient in real-world situations, usually due restrictions imposed by one’s cohabitee.

In the BeoLab 5, B&O have aimed to make a design with a forward directivity similar to that of a dipole but, due to the attenuated rear-response, one which can be placed directly against a wall.

Taking the power average from nine measurements made at random room positions, yields some kind of loudspeaker power response. Other measurement techniques have been tried, but this power averaging technique, Gert reports, shows better correlation with subjective testing.

Efficiency of loudspeakers is generally low and the BeoLab 5 is no exception. Free-field, 200W of electrical power input might yield 1W of acoustic power. The BeoLab 5 contains amplifiers capable of supplying around 2.5kW of power!

Gert notes there can be huge changes in power response at around 100Hz for differing speaker placement, so a filter is introduced to equalise the power response positioning-room. A normal tone control can never compensate for this kind of problem; much more precise control is provided in the BeoLab 5 using Digital Signal processing.

The BeoLab 5 includes a formidable array of signal processing. The crossovers are performed digitally, and much more besides.

During factory test, the response of each driver is automatically equalised to compensate for manufacturing tolerances. Overall equalisation is also applied to achieve the overall target frequency response. This production testing employs a total of 6 microphones – four at the front (one close to each of the drivers) and two at the rear. A reference speaker provides the target for the equalisation process . Each production speaker is adjusted to match the frequency response of this reference unit with a target error of less than 0.5dB.

Temperature and air pressure can alter the measurements significantly, so these are monitored during this phase.

Using an in-built, motorised microphone which slides out from under the speaker, automatic correction of low-frequency response up to around 300Hz can be invoked by the user to reduce the effects of the room in which the speakers are placed. Gert points out that this correction is not a modal correction – it’s more like a general equalisation, with the filter response being smoothed during the measurement process.

Interestingly, the target response for this ‘auto-correction’ system is not, as one might expect, a flat response, but rather a response that has been determined empirically through critical listening.

The thermal monitoring uses a combined technique of feed-forward modelling in conjunction with average temperature measurement of the driver mechanical assembly. Each driver also has thermal modelling, arranged such that should, on average, too much power be applied to any driver, progressive attenuation is applied to its output (and also to outputs to all drivers of higher frequency to maintain a consistent tonal balance).

A “party test” is also carried out which runs the speakers at full-power for three days!

The BeoLab 5 is a no-compromise design that might at first appear to be at the more esoteric end of hi-fi. But many thousands of units have been sold, proving that many consumers still aspire to achieve great audio reproduction and are prepared to buy-in to new technology to achieve it.

It was fascinating to hear about the design philosophy and gain some insight into the processes. On behalf of all present I’d like to extend thanks both to Gert for the presentation, and to B&O for making it possible.

Report by Nathan Bentall

Conducted by Keith Howard

Bob Stuart has been a major figure in the British audio industry for over 30 years. Best known as Chairman and co-founder, with Allen Boothroyd, of what is today Meridian Audio Ltd, he has done much more than steer the company through challenging times to its current high-profile position manufacturing some of the most sophisticated audio equipment available. A pioneer of active and then DSP-equipped loudspeakers, he was quick to recognise the potential of CD and, as part of the ARA, to push for a version of DVD dedicated to high-resolution multichannel audio. Meridian’s own lossless compression algorithm, MLP, was developed in anticipation of this and selected by the DVD Forum for DVD-Audio in a technology shoot-out against stern competition. In expanded form it remains the basis of the Dolby TrueHD lossless compression scheme used in Blu-ray Disc. With a long-standing interest in psychoacoustics, which he studied alongside electronic engineering at Birmingham University, Bob is one of very few creators of high-quality audio equipment to have explored the fundamentals of sound perception and generated computer models of human hearing to help guide the design process. In recent years, in collaboration with Peter Craven, he has investigated the effects of digital anti-aliasing and reconstruction filters, one intriguing result being that Meridian’s latest flagship CD player – the 808.2 Signature Reference – uses minimum-phase rather than linear-phase output filtering.

These subjects and many others are covered in this interview, with Bob presenting supporting material to clarify the issues.

An Interview with Bob Stuart (audio, 23MB)

Title: ‘Surround sound audio codecs in broadcasting – an introduction and latest results from independent listening tests’
Location: The Royal Academy of Engineering, London
Description: Lecture by David Marston, BBC R&D
Start Time: 18:30
Date: 14th April 2009

Abstract

Surround sound systems are now becoming a popular addition to many people’s homes. This means there is now a demand for surround sound content to be delivered to homes via broadcasting, Internet or recorded media. Whichever way it gets to its destination, it is going to require data reduction along its journey. This may be in the transmission end of a broadcast chain, or in the transport of audio from a studio out over a broadcaster’s network.

This data reduction uses audio coders designed for surround sound. There are currently numerous different audio coders available, often with different attributes and performance. Choosing which coder to use is not a simple choice, and one of the key factors in this choice is the sound quality. It is inevitable that for serious data reduction, the coder will have to be lossy and therefore compromise sound quality. Our work assessed the sound quality of a selection of audio coders using the most accurate instrument of measurement available: the human ear. Here we present the codecs tested, how the tests were done, and of course the results.

Meeting Report

This paper described the methodology used for a series of evaluation tests conducted by members of the EBU on a range of commercially available audio codecs.

In his introduction, David explained that the measurement of perceptual audio coding systems cannot be carried out using conventional objective measuring tools, as one would do for wow and flutter, for example. An objective measure based on psychoacoustic principles such as PEAQ can work reasonable well with MPEG-style stereo codecs, but there is nothing available yet for surround systems. A disadvantage of using such measurement is that any new method is likely to be incorporated into a codec’s design to ensure good test results.

The only effective test, therefore, is subjective listening using humans – a slow and expensive process if a good sized sample is employed, although you do get useful results.

There are various parameters that can be looked at: overall quality, spatial quality in the case of surround sound, intelligibility, cascaded codecs, and so on. When a selection of different codecs and coding rates have to be tested in multiple combinations, the complexity increases further. In these instances a measurement system such as PEAQ can be used as a pre-filter.

The main subjective testing methods today are MUSHRA (MUlti Stimulus test with Hidden Reference and Anchors); BS1534, which is designed for mid-range to higher quality codecs, can test multiple codecs at the same time and was used in the EBU tests; BS1116, designed for high quality codecs but only samples one at a time; and P800. The latter is for speech and was not relevant for these tests.

MUSHRA produces a quality value and BS1116 an impairment value for each codec. On occasions it may be relevant to have more than one value, for example for temporal and spatial quality. A single value makes testing faster, as well as being easier for the listener and for analysis, however it can hide differences in listeners’ perceptions.

Ensuring a gender balance has also been a problem as most of the listeners have been male. Training is important, whoever takes the tests. Listeners must be taught to identify coding artefacts and other problems, as well as how to use the assessment interface. For scoring, a numerical scale is useful because it avoids interpretations of words like ‘Fair’ or ‘Good’.

Each listener hears five codecs, any more would make the test too tiresome and could degrade the accuracy of results. During the MUSHRA test listeners are always given the reference, and also included in the randomised sequence is a hidden low quality anchor reference, a 3.5kHz low-pass filtered version of the original. In the EBU test another, spatially reduced, anchor was added. For BS1116, listeners hear one codec at time, which is compared with a hidden reference and the known reference. This takes much longer, therefore each listener is limited to four codecs.

It is important to select a cross-section of experienced and novice listeners. Some may prove to have poor listening skills, or have a hearing impairment, but it is not always possible to identify this in advance. So it is better to use them for the test and reject their findings afterwards, often based on their ability to rank the hidden reference and low quality anchor.

David showed a slide of the MUSHRA test interface and explained how the listener can select each of the examples in order to make direct comparisons with the reference. He went on describe listening set-up at Kingswood Warren – soon to disappear!

Choosing the test material is always difficult. It must be critical, in order to highlight coding artefacts, but at the same time be unbiased, eg not material that is known to disadvantage a specific codec. The material must also be appropriate for the application: a mixture of music, speech and jingles (which will already have been compressed) for a broadcast codec, for example. The final choice of ten pieces of test material was made by a selection panel.

One of the techniques used by Institut für Rundfunktechnik (IRT), which analysed the results, was the Spearman Rank Correlation. This looks at the ranking of all the scores, and if anybody’s ranking was massively different from the average they were rejected. Around ten percent of listeners were eliminated at this stage.

There are three phases to this series of tests. The first two covered the most commonly used codecs for emission (transmission), the last link in the chain and usually the one with the lowest bit rate. Phase three looked at combinations of higher bit rate codecs used in the production/distribution chain – which are designed to be cascaded – combined with low bit rate emission codecs, and how they interact.

To ensure randomisation it was decided to split the codecs into three groups, based on their bit rates. Each listener’s five codecs contained at least one from each of these high, medium and low bit-rate groups, with the remaining two being from a single group to ensure a strong intra-group comparison within the test; eg a listener might hear one high, three medium and one low bit-rate codec.

Ten test items were used covering a varied selection of material, including applause, harpsichord, sax and piano, a church organ and Robert Plant.

IRT carried out the analysis to produce the test results. Some listeners were rejected if they fell outside of the Spearman Rank Correlation threshold, which compares the ranking given by each listener with the overall rankings. After this process some codecs dropped below the minimum of 15 listeners and so extra listening tests had to be carried out.

David went on to show the various test results and explained that some of the codecs used for the test were pre-production prototypes, or have since been upgraded. One common element was that the most difficult item to encode – usually the applause – normally ranked much lower than the mean. For example, one codec was rated 30 on applause but 90 on music, proving that perceptual coding is very content-dependent. [Note: This report does not list the codecs involved or their rankings due to the risk of misrepresenting the current performance of those codecs.]

The conclusion from Phase 1, as would be expected, was that higher bit rates produce better quality. The detailed results for each codec have hopefully given their developers something to work on in terms of improving their performance.

Phase 2 retained the applause sample from Phase 1 as a reference item but the other samples, although similar in terms of content type, were different. When results from Phases 1 and 2 were compared they were similar, proving that the testing methodology was valid. Phase 2 again showed that excellent quality can be achieved from low bite-rate codecs, but not for every type of content, and again it gave the developers guidance on areas where improvements can be made.

Phase 3 combined cascaded high bit-rate distribution codecs such as Dolby E, apt-x and Linear Acoustics with a selection of emission codecs. Ten items were selected from the samples used in the previous tests and these were cascaded five times through the same distribution codec before being passed through one or two different emission codecs. Various combinations were tested.

It was decided to use BS1116 rather than MUSHRA for this phase. Because this is an impairment scale, it was not possible to make any direct comparisons with the results of Phases 1 and 2. The conclusion was that distribution codecs still introduce some impairment, having the effect of creating a ‘ceiling’ to the overall quality attainable. The recommendation therefore is to use the highest bit rate possible.

Overall conclusions from these listening tests were that perceptual coding is still an imperfect art and there is room for improvement. Analysis is not easy, but these tests do reveal things that objective tests could never do, as well as uncovering things you wouldn’t expect.

Meeting report by Bill Foster