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Orban Loudness Meter Public Beta
The CBS meter is a “short-term” loudness meter that displays the details of moment-to-moment loudness with dynamics similar to a VU meter. It uses the Jones & Torick algorithm developed at the CBS Technology Center [Bronwyn L. Jones and Emil L. Torick, “A New Loudness Indicator for Use in Broadcasting,” J. SMPTE September 1981, pp. 772-777]. Created using Orban-developed modeling software, the DSP implementation typically matches the original analog meter within 0.5 dB on sinewaves, tone bursts and noise. The Jones & Torick algorithm improves upon the original loudness measurement algorithm developed by CBS researchers in 1967. Its foundation is psychoacoustic studies done at CBS Laboratories over a two year period by Torick and the late Benjamin Bauer, who built on S. S. Stevens’ ‘50s-era work at Harvard University. After surveying existing equal-loudness contour curves (like the famous Fletcher-Munson set) 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 semireverberant 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 Measurement,” IEEE Transactions on Audio and Electroacoustics, Volume AU-14, Number 3, September 1966, pp. 141-151, along with results from other tests whose goal was to model the loudness integration time constants of human hearing. These studies concentrated on the moderate sound levels typically preferred by people listening to broadcasts (60 to 80 phons) and did not attempt to characterize loudness perception at very low and high levels. (The phon is a unit of perceived loudness, equal in number to the intensity in decibels of a 1 kHz tone judged to be as loud as the sound being measured. [The American Heritage® Science Dictionary, 2005]) According to this research and its predecessors, the four most important factors that correlate to the subjective loudness of broadcasts are these: 1. The power of the sound. 2. The spectral distribution of the power. The ear’s sensitivity depends strongly on frequency. It is most sensitive to frequencies between 2 and 8 kHz. Sensitivity falls off fastest below 200 Hz. 3. Whether the power is concentrated in a wide or narrow bandwidth. For a given total sound power, the sound becomes louder as the power is spread over a larger number of critical bands (about 1/3 octave). This is called loudness summation. 4. Temporal integration: As its duration increases, a sound at a given level appears progressively louder until its duration exceeds about 200 milliseconds, at which point no further loudness increase occurs. Bauer and Torick used the results of this research to create a loudness meter with eight octave-wide filters, each of which covers three critical bands. (B & T did not use one filter per critical band because this would have made the meter, which was realized using analog circuitry, prohibitively expensive.) Each filter feeds a full-wave rectifier and each rectifier feeds a nonlinear lowpass filter that has a 10 ms attack time and a 200 ms release time, somewhat like the sidechain filter in an AGC. This models the “instantaneous loudness” perception mechanism in the ear. Instantaneous loudness is not directly perceived but is an essential part of the total loudness model. To map the instantaneous loudness to perceived short-term loudness, the outputs of each of the nonlinear lowpass filters are arithmetically summed with gains chosen to follow the 70 phon equal-loudness curves of the ear. The sum is applied to a second, slower nonlinear lowpass filter. This has an attack time of 120 ms and a release time of 730 ms. Along with the eight nonlinear lowpass filters following the individual filters, this filter models temporal integration and maps it to the visual display. Meanwhile, the arithmetic addition models loudness summation. The accepted unit of subjective loudness is the sone. With a sinewave, 40 phons = 1 sone. A doubling of sones corresponds to a doubling of loudness. However, because broadcasters were accustomed to working in decibel units, J & T chose to map loudness on a display encompassing –20 to +5 dB in 0.5 dB increments, with the understanding that the perceived loudness doubles every 10 dB at loudness levels typically heard by broadcast audiences. A reasonable calibration level is 0 dB = 75 phons = 11.3 sones. The J & T meter is monophonic. Psychoacoustic studies indicate that when multiple acoustic sources are present in a room, loudness is most accurately expressed by summing the power in the sources: Driving two loudspeakers with identical program produces 3 dB higher loudness than a single speaker produces. Therefore, to extend the J & T algorithm to multichannel reproduction, we implement one eight-filter filterbank for each channel and compute RMS sums of the outputs of corresponding filters in each channel before these sums are applied to the eight nonlinear lowpass filters. As in the monophonic J & T algorithm, the sum of these lowpass filters drives a second nonlinear filter, which drives the display. Our software version of the meter has a 30 dB range with graphic resolution of approximately 0.1 dB.
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Experimental Long-Term Loudness Indication from the CBS Meter We have added an experimental long-term loudness indication by post-processing the J & T algorithm’s output. Displayed by a single cyan bar on the “CBS” meter, this uses a relatively simple algorithm and we welcome any feedback on its perceived usefulness. (We expect to refine it in the future.) This algorithm attempts to mimic a skilled operator’s mental integration of the peak swings of a meter with “VU-like” dynamics. The operator will concentrate most on the highest indications but will tend to ignore a single high peak that is atypical of the others. The long-term loudness algorithm displays the average of the peak indications of the meter over a user-determined time period. The average is performed before dB conversion. All peak indications within the time period are weighted equally with the following exceptions: --If the maximum peak in the window is more than 3 dB higher than the second highest peak, it is discarded. --All peaks more than 6 dB below the maximum (or second-to-maximum, if the maximum peak was discarded) are discarded. The ITU BS.1770 Meter In 2006, the ITU-R published Recommendation ITU-R BS.1770: “Algorithms to measure audio programme loudness and true-peak audio level.” Developed by G.A. Soulodre, the BS.1770 loudness meter uses a frequency-weighted RMS (power summation) measurement intended to be integrated over several seconds—perhaps 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 integration time constants of human hearing, as does the CBS meter. Orban’s BS.1770 loudness meter uses the Leq(RLB2) algorithm as specified in the Recommendation. This applies frequency weighting before the RMS integrator. The frequency weighting is a series connection of pre-filter and RLB weighting curves; there is one set of filters for each stereo channel, as in the CBS meter. The output of each filter set is squared and then summed into the integrator. To indicate long-term loudness, the meter displays the square root of the integrated value on a dB scale. The Orban meter precisely implements equations (1) and (2) in this document by using a rolling integrator whose integration time is user-adjustable from one to ten seconds. This integrator equally weights all samples from the present time to x seconds before the present time, where x is the integration time set by the user. To do this, the sample that is x seconds old is subtracted from the integrator before the new, present-time sample is added. This requires a substantial amount of memory because all samples from the present time to x seconds earlier must be retained. Meter Scales In their original publications and standards, each of the meters implemented in the Orban Loudness Meter has a different specified scale and range. To best allow users to compare the indications of the various meters under dynamic program conditions, we chose to present their indications on identical linear-dB scales extending from 0 to –30 dB with respect to digital full-scale. The CBS and VU meters have gain adjustments that allow users to choose their preferred lineup level. Conformance with Published Standards Our implementation of the PPM can be switched for 5 ms or 10 ms attack times because there are standards for both variations. The “10 ms attack” mode follows EBU Tech. 3205-E as closely as possible given the difference in the scale and the limitations introduced by the Orban meter’s 48 kHz internal sample rate. In practice, this means that its indication obeys the dynamic performance specification of the standard within 0.5 dB for tone burst durations of 100, 10, and 5 ms. Because of undersampling, the Orban PPM under-reads a 5 kHz 1.5 ms burst by about 3 dB and a 10 kHz 0.5 ms burst by about 4 dB compared to the standard. In a future version of the meter, we may oversample its detector to comply more closely with the 1.5 ms and 0.5 ms specifications. Our implementation of the VU meter reaches 99% (–0.09 dB) of steady-state when presented with a 1 kHz tone burst with an “on” duration of 300 ms and an “off” duration of 500 ms or more . In concordance with the standard, the meter has an overshoot of 1%. Because its reading is presented on a dB-linear scale instead of a standard VU “A” or “B” scale, we believe that this is the closest we could come to the spirit of this meter. Our true peak-reading meter reads the peaks of the 48 kHz digital samples within the meter. It does not attempt to extrapolate the peaks of the signal after D/A conversion, as specified in the BS.1770 standard. This requires oversampling the peak detector, which we may do in a future release. Also, note that the Orban meter may not indicate the true peak sample values of material not originally at 48 kHz sample rate. Windows will sample-rate-convert such material to 48 kHz before the meter and will change the values of the samples when it does so.
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