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dBFS

dBFS, or decibels relative to full scale, is a unit of measurement used to express the amplitude levels of digital audio signals in systems such as pulse-code modulation (PCM), where levels are referenced to the maximum representable value in the digital domain.^[1] According to the Audio Engineering Society (AES) standard AES17, 0 dBFS is defined as the level of a sine wave whose peak amplitude reaches the largest positive digital code value, representing the absolute maximum signal level before digital clipping occurs.^[2] In digital audio, dBFS values are always negative or zero, with signals below full scale expressed as, for example, -20 dBFS, indicating a reduction in amplitude relative to 0 dBFS.^[3] This scale provides a fixed reference for measuring peak and RMS levels, enabling precise gain staging and loudness assessment in recording, mixing, and playback workflows.^[4] Unlike absolute units, dBFS is dimensionless and specific to digital systems, where the full-scale limit is determined by the bit depth—for instance, 16-bit audio offers a theoretical dynamic range of approximately 96 dB from 0 dBFS down to the noise floor.^[1] dBFS is distinct from analog audio measurement units like dBu and dBV, which reference voltage levels rather than digital quantization. For example, 0 dBu corresponds to 0.775 volts RMS (a legacy reference from early telephone standards), commonly used in professional audio equipment, while 0 dBV equals 1 volt RMS, typical in consumer gear. Conversions between dBFS and these analog units depend on the interface's design, such as digital-to-analog converters (DACs); professional interfaces align analog reference levels to digital scales differently by standard—for example, the EBU R68 sets -18 dBFS to 0 dBu, while SMPTE RP155 sets -20 dBFS to +4 dBu, allowing headroom before 0 dBFS clipping.^[5]^[6] The adoption of dBFS has been integral to standards for loudness normalization, such as those in AES and ITU recommendations, ensuring consistent audio levels across streaming, broadcasting, and mastering applications.^[1]

Fundamentals

Definition

dBFS, or decibels relative to full scale (dB FS), is a logarithmic unit used to express the amplitude levels of signals in digital systems, particularly in digital audio encoded via pulse-code modulation (PCM).^[1] It measures the signal relative to the maximum possible digital value, where 0 dBFS corresponds to the clipping point—the highest amplitude before digital overflow occurs, represented by all bits set to 1 in the binary code (or the equivalent maximum code value in signed representations).^[7] This reference distinguishes dBFS from absolute decibel scales, as "FS" denotes full scale, the inherent capacity of the digital format without tying to physical units like voltage or power.^[8] The amplitude-based formula for dBFS is given by

\text{dBFS} = 20 \log_{10} \left( \frac{x}{x_{\max}} \right),

where x is the instantaneous or peak signal amplitude and x_{\max} is the full-scale amplitude.^[7] For power measurements, the formula adjusts to

\text{dBFS} = 10 \log_{10} \left( \frac{P}{P_{\max}} \right),

where P is the signal power and P_{\max} is the maximum power corresponding to full scale; notably, standards like AES17 define 0 dBFS specifically as the root-mean-square (RMS) value of a full-scale sine wave with peak amplitude at 100% full scale.^[1] In fixed-point digital representations common to audio, such as 16-bit or 24-bit PCM, signals are quantized to discrete integer levels, with full scale typically defined as $2^{n-1} - [1](/page/1) for n-bit signed integers (e.g., 32767 for 16-bit).^[7] This quantization imposes a finite resolution on the dBFS scale, limiting the smallest distinguishable levels near the noise floor and influencing the overall dynamic range achievable in the system.^[9]

Relation to Other Decibel Scales

dBFS measures signal levels relative to the full-scale amplitude of a digital system, where 0 dBFS represents the maximum possible digital value without clipping, lacking any absolute reference to physical quantities like voltage or power.^[3] In contrast, analog decibel scales such as dBu and dBV are defined with respect to specific voltage levels: dBu references 0.775 volts RMS, derived from the voltage producing 1 milliwatt in a 600-ohm load, while dBV references 1 volt RMS, both independent of load impedance in modern usage.^[10] Similarly, dBm is an absolute power scale relative to 1 milliwatt, originally assuming a 600-ohm impedance, making it suitable for electrical power measurements but not directly applicable to digital domains.^[10] Direct conversion between dBFS and these analog scales is not possible without knowledge of the digital-to-analog converter's (DAC) reference level, as dBFS is system-specific and tied solely to the digital word length, whereas analog scales depend on hardware-defined voltage or power outputs.^[3] For instance, in professional audio interfaces, a common alignment sets 0 dBFS to correspond to +18 or +24 dBu at the analog output, but this varies by device calibration and cannot be universally applied.^[3] A frequent misconception arises from equating dBFS with dBSPL (decibels sound pressure level), which measures acoustic pressure relative to 20 micropascals—the approximate threshold of human hearing—making it suitable for environmental or perceptual loudness assessments but irrelevant to digital signal processing.^[11] dBFS, being confined to the digital domain, cannot represent acoustic measurements without additional transduction and calibration through microphones or speakers, rendering such direct comparisons invalid.^[12] In hybrid analog-digital workflows, dBFS interfaces with other scales primarily at the points of analog-to-digital (ADC) or digital-to-analog (DAC) conversion, where reference levels are aligned to prevent distortion or mismatch; for example, professional line levels around +4 dBu often map to -18 dBFS in calibrated systems to maintain headroom.^[3] This alignment ensures seamless signal flow but requires careful monitoring to avoid assuming equivalence across domains.^[10]

Amplitude Measurements

Peak Levels

Peak dBFS refers to the measurement of the highest instantaneous sample value in a digital audio waveform, expressed in decibels relative to full scale, where the maximum possible value is 0 dBFS corresponding to the largest representable digital code.^[2] This scale ensures that all peak levels are at or below 0 dBFS, preventing overflow in fixed-point digital systems. For a sinusoidal waveform, the peak dBFS is calculated by taking the 20-log10 ratio of the maximum sample amplitude to the full-scale amplitude, aligning with the general dBFS formula but focused on the absolute peak samples rather than average power. A full-scale sine wave, where the peaks reach the maximum digital code, thus measures 0 dBFS at its peaks, with its root-mean-square (RMS) value at -3 dBFS due to the 3 dB relationship between peak and RMS for sine waves.^[2] In audio metering, distinguishing between sample peak and true peak is essential, as sample peak only considers discrete sample values while true peak accounts for potential higher amplitudes between samples during digital-to-analog conversion. True peak metering involves oversampling the signal—typically by a factor of 4 or higher—to detect inter-sample peaks, which can exceed 0 dBFS and lead to aliasing distortion or clipping in playback systems if not addressed. Standards like ITU-R BS.1770 define true peak as the maximum level in the reconstructed continuous-time waveform, recommending its use to ensure compliance with delivery specifications.^[13] Levels approaching 0 dBFS in peak measurements trigger hard clipping, where digital samples are truncated, introducing harsh distortion artifacts. To accommodate transients and processing, a recommended headroom of -6 dBFS or greater for peak levels is advised during mixing, providing buffer against inter-sample overs and subsequent gain boosts.^[14] In audio production, digital audio workstations (DAWs) employ peak meters to monitor these levels in real-time, alerting users to potential overs and enabling adjustments via gain staging to maintain signal integrity throughout the workflow.^[15]

RMS Levels

The root mean square (RMS) level in dBFS provides a measure of the average power or sustained loudness of a digital audio signal, calculated relative to full scale. The RMS value of a signal x consisting of N samples is given by \mathrm{RMS}(x) = \sqrt{\frac{1}{N} \sum_{i=1}^{N} x_i^2}, and the corresponding dBFS level is $20 \log_{10} \left( \frac{\mathrm{RMS}(x)}{x_{\max}} \right), where x_{\max} is the maximum possible digital level (typically normalized to 1). This formulation ensures that a full-scale square wave, where all samples reach x_{\max}, yields an RMS level of 0 dBFS, representing the theoretical maximum average power in the digital domain. Unlike peak levels, which capture instantaneous maxima, RMS levels reflect the effective energy over time, making them suitable for assessing perceptual loudness. For a sinusoidal waveform with a peak at 0 dBFS, the RMS level is approximately -3 dBFS, since the RMS of a sine wave is its peak amplitude divided by \sqrt{2}, yielding $20 \log_{10} (1 / \sqrt{2}) \approx -3.01 dBFS. This -3 dB difference highlights how RMS averages the signal's varying amplitude, providing a lower value than the peak for periodic waveforms like sines. For non-sinusoidal signals, such as square waves at full scale, the RMS aligns directly with the peak at 0 dBFS due to constant amplitude. In digital systems, RMS calculations employ the true mathematical definition, computing the square root of the mean of squared sample values across a defined window (e.g., short-term or integrated over the program). This contrasts with approximations in some analog or real-time hardware meters, which may use simplified ballistics or scaling factors for responsiveness but can introduce minor inaccuracies for complex waveforms. True RMS ensures precise handling of arbitrary shapes, such as music with high crest factors, where the ratio of peak to RMS (in dB) quantifies dynamic range—typically 6–20 dB for audio content.^[1] RMS levels are integral to audio metering for broadcast and production compliance, often integrated into standards like ITU-R BS.1770 for loudness units relative to full scale (LUFS), which builds on weighted RMS measurements to normalize program loudness. Program meters display RMS to monitor sustained levels, ensuring adherence to targets such as -23 LUFS for European broadcasting under EBU R 128, derived from RMS-like averaging. The crest factor, computed as peak dBFS minus RMS dBFS, aids in evaluating signal dynamics and preventing over-compression.^[1] In practice, average music levels are often maintained at -12 to -20 dBFS RMS to preserve headroom against peaks, aligning with broadcast alignment signals at -18 dBFS for a 1 kHz tone (equivalent to its RMS). This range allows for transients up to 0 dBFS without clipping while supporting consistent loudness across playback systems.^[16]

Applications

Dynamic Range

In digital audio systems, dynamic range measured in dBFS represents the span from the maximum undistorted level at 0 dBFS to the noise floor, quantifying the system's ability to capture both loud and quiet signals without excessive distortion or inaudibility. This range is primarily determined by the quantization noise inherent in the analog-to-digital conversion process, where the noise floor for an n-bit system is approximately -6n dBFS, yielding a theoretical dynamic range of about 6n dB.^[17] For instance, 16-bit audio, as used in compact discs (CDs), provides a theoretical signal-to-noise ratio (SNR) of 96 dB, with the noise floor at -96 dBFS. Bit depth is the primary factor influencing this dynamic range, as each additional bit effectively adds 6 dB of resolution, extending the usable span below full scale. A 24-bit system achieves approximately 144 dB of dynamic range, allowing for much quieter signals to be represented faithfully relative to 0 dBFS.^[18] Dithering techniques further enhance the effective range by adding low-level noise to randomize quantization errors, enabling the capture of signals up to 1-2 bits below the nominal floor without introducing audible distortion, particularly when reducing bit depth during processing or export. Additionally, higher sampling rates facilitate noise shaping in oversampled converters, redistributing quantization noise to ultrasonic frequencies outside the audible band, thereby improving the perceived dynamic range within the 20 Hz to 20 kHz range by up to several dB depending on the oversampling ratio.^[19] In practice, the achievable dynamic range in dBFS is often limited below theoretical values due to analog noise introduced by converters and other components in the signal chain. High-quality audio ADCs and DACs may exhibit self-noise floors around -120 dBFS or higher in 24-bit systems, but environmental factors and analog circuitry can raise this to -100 dBFS or more, constraining the overall range.^[20] Quiet passages in recordings, such as ambient sounds or instrument decays, are particularly susceptible to this self-noise, which can mask subtle details if signals dip too close to the floor.^[21] In recording applications, dBFS dynamic range guides level management to ensure signals remain well above the noise floor while avoiding overload at 0 dBFS. For vocals, typical quiet elements are maintained at least -60 dBFS to preserve clarity over the noise floor, allowing natural dynamics without undue amplification of hiss.^[22] Dynamic range compression is commonly applied to fit wide-ranging source material within this span, reducing the difference between peaks and troughs to better utilize the available headroom in digital formats like CDs, where the 96 dB theoretical range sets a benchmark for SNR performance.^[23]

Headroom and Clipping

Headroom in digital audio refers to the margin between the average root mean square (RMS) level of a signal and the maximum level of 0 dBFS, allowing transients to occur without exceeding the digital ceiling and causing distortion.^[14] In mixing workflows, this margin is typically maintained at 6-12 dB to accommodate processing effects and preserve dynamic flexibility.^[14] Clipping occurs when an audio signal surpasses 0 dBFS, resulting in hard digital clipping where the waveform peaks are abruptly truncated, introducing predominantly odd-order harmonics that produce harsh, unpleasant distortion.^[24] To mitigate this, soft limiting techniques apply gradual gain reduction, emulating analog saturation to control peaks while minimizing unwanted artifacts.^[24] Best practices in mastering involve targeting peak levels at -1 to -3 dBFS to ensure safety during final output and distribution.^[25] Additionally, inter-sample peak monitoring is crucial in playback chains, as reconstruction filters in digital-to-analog converters can generate overshoots beyond sample peaks, potentially leading to clipping if not anticipated.^[26] The consequences of clipping include audible artifacts like crackling and metallic harshness that degrade listening quality.^[27] In fixed-point digital systems, such as standard 16- or 24-bit PCM formats, clipping causes irreversible data loss by capping values at the maximum representable level, preventing accurate signal recovery.^[28] In live sound reinforcement, adequate headroom is essential to prevent system overload during sudden volume increases, such as crescendos in orchestral performances.^[29] Headroom requirements are closely tied to a signal's crest factor—the ratio of its peak level to RMS level—with high-crest-factor sources like drums demanding more margin (often 10-15 dB or greater) to handle sharp transients without distortion.^[30]

Comparisons

Analog Level Equivalents

In professional audio environments, the mapping between dBFS and analog levels is designed to provide sufficient headroom while aligning nominal operating levels for compatibility between digital and analog equipment. Commonly, 0 dBFS corresponds to +24 dBu, representing the maximum analog level just before clipping in many studio interfaces and converters, ensuring that digital full scale does not immediately overload analog stages.^[4] The nominal line level of +4 dBu is typically aligned to -20 dBFS in North American broadcast standards per SMPTE RP155, providing 20 dB of headroom above alignment to accommodate peaks.^[31] In contrast, the European Broadcasting Union (EBU) R68 standard aligns 0 dBu to -18 dBFS, with the professional nominal +4 dBu thus at -14 dBFS, emphasizing 18 dB of headroom for production workflows.^[5]^[32] For consumer audio systems, such as CD players and home hi-fi equipment, 0 dBFS typically corresponds to 2 V RMS (+6 dBV). The standard nominal line level of -10 dBV (0.316 V RMS) thus aligns to approximately -16 dBFS in many devices, providing about 16 dB of headroom, though this can vary (e.g., -22 dBFS in some interfaces).^[33] These mappings ensure that digital sources like CD players, which output up to +6 dBV at 0 dBFS, integrate smoothly with analog consumer chains.^[34] Mismatches in these alignments can lead to suboptimal performance, such as signals appearing too low in level (requiring excessive makeup gain and introducing noise) or causing distortion if digital peaks exceed analog headroom during conversion. Proper calibration of audio interfaces is essential, often involving test tones at the specified alignment levels to match digital and analog domains. For instance, in recording studios, where 0 dBFS may surpass the +24 dBu analog limit, careful gain staging—keeping peaks below -6 dBFS—prevents clipping while preserving dynamic range across hybrid digital-analog workflows.^[33]^[35]

Voltage and Power Conversions

To convert a dBFS level to an absolute voltage, the full-scale voltage V_{fs} of the specific digital-to-analog converter (DAC) must be known, as dBFS is a relative digital scale without a fixed absolute reference. The conversion to dBV (decibels relative to 1 V RMS) is given by the formula:

\text{dBV} = \text{dBFS} + 20 \log_{10} \left( \frac{V_{fs}}{1 \, \text{V}} \right)

This derives from the standard decibel voltage ratio, where the absolute voltage V is V = V_{fs} \times 10^{(\text{dBFS}/20)}, and dBV follows $20 \log_{10} (V / 1 \, \text{V}). For instance, if V_{fs} = 2 \, \text{V RMS}, a signal at -6 dBFS corresponds to $2 \times 10^{-6/20} \approx 1.00 \, \text{V RMS}, or +0 dBV.^[36] Power conversions from dBFS require both V_{fs} and the system impedance Z, as power P = V^2 / Z. The full-scale power is P_{fs} = V_{fs}^2 / Z, and the conversion to dBm (decibels relative to 1 mW) uses:

\text{dBm} = \text{dBFS} + 10 \log_{10} \left( \frac{P_{fs}}{0.001 \, \text{W}} \right)

Here, the absolute power P is P = P_{fs} \times 10^{(\text{dBFS}/10)}, with dBm as $10 \log_{10} (P / 0.001 \, \text{W}). For example, assuming Z = [600](/page/600) \, \Omega and V_{fs} = 12.28 \, \text{V RMS} (corresponding to +24 dBu full scale in professional setups), P_{fs} \approx 0.251 \, \text{W}, so 0 dBFS equals +24 dBm, and -10 dBFS equals +14 dBm. These calculations assume sinusoidal signals and RMS values for consistency with audio power metrics.^[36] In professional audio interfaces aligned to +4 dBu nominal levels, 0 dBFS typically equates to +24 dBu, corresponding to approximately 12.28 V RMS or 12.28 V peak-to-peak in balanced systems (accounting for differential signaling). This provides 20 dB of headroom above the +4 dBu reference (1.228 V RMS). For consumer -10 dBV systems, full-scale output is often standardized at 2 V RMS for 0 dBFS in devices like CD players and home DACs, yielding +6 dBV and roughly 16 dB headroom above the -10 dBV nominal level (0.316 V RMS). These equivalents vary by manufacturer specifications, such as DAC output stages.^[37]^[36] Such conversions are inherently system-dependent and not universal, as they rely on precise DAC or ADC specifications like V_{fs} and [Z](/page/Z), which differ across equipment (e.g., balanced vs. unbalanced outputs). Without these details, absolute values cannot be determined solely from dBFS. In practice, audio production tools like digital audio workstations (DAWs) or hardware meters (e.g., those from RME or Focusrite interfaces) perform real-time conversions by incorporating device calibration profiles, enabling monitoring in mixed dBFS and absolute units during mixing or mastering.^[36]

Historical Development

Origins

The concept of dBFS emerged during the 1970s and early 1980s as part of pulse-code modulation (PCM) research for digital audio, with early work at institutions like Bell Labs and NHK focusing on defining full-scale signal levels relative to the maximum digital code value to manage quantization and headroom. Bell Labs' foundational PCM telephony developments from the 1930s influenced later digital audio efforts. Meanwhile, NHK advanced practical PCM recorders starting in 1967 with a monophonic system at 12-bit resolution and 30 kHz sampling, upgrading to a stereo version in 1969 with 13-bit resolution at 32 kHz.^[38]^[39] A pivotal milestone came in 1982 with the introduction of the Compact Disc format by Philips and Sony, which standardized 16-bit PCM audio where 0 dBFS represented the maximum unclipped level, yielding a theoretical dynamic range of 96 dB based on the bit depth's quantization steps.^[40] This specification, detailed in the Red Book standard, established the full-scale reference for consumer digital audio playback. The term "dBFS" first appeared in print around 1977, building on earlier uses of "dB below full scale" since the 1950s.^[41] The adoption of dBFS drew from analog audio practices, adapting the established decibel scale—rooted in logarithmic ratios to align with human hearing sensitivity—to digital quantization, ensuring that level measurements reflected perceptual loudness rather than linear amplitude.^[42]

Standardization and Adoption

The formal standardization of dBFS measurement emerged in 1991 with AES17, which defined 0 dBFS as the RMS level of a full-scale sine wave, providing methods for digital audio performance verification. The Audio Engineering Society also published AES3-1985, defining the serial interface for two-channel PCM audio transmission and implying maximum digital levels to avoid overflow.^[2]^[43] In 1987, the International Electrotechnical Commission issued IEC 60908 for the Compact Disc Digital Audio system, specifying 16-bit linear PCM encoding with full-scale amplitude, ensuring consistent playback without distortion across compliant devices.^[44] Building on these foundations, the European Broadcasting Union released Recommendation R68 in 2000, establishing -18 dBFS as the alignment level for digital audio in broadcast production to provide adequate headroom.^[5] Adoption of dBFS accelerated in professional and consumer audio during the 1990s and 2000s. Digital audio workstations integrated dBFS metering as the default scale for monitoring and mixing, enabling precise control in 16-bit and emerging 24-bit environments. By the 2000s, dBFS became standard in compressed consumer formats like MP3. In the 2010s, streaming services adopted loudness normalization frameworks integrating dBFS-based true-peak limits with LUFS measurements, as outlined in ITU-R BS.1770 (first published in 2006 and revised through 2023), to ensure consistent volume across platforms like Spotify and Apple Music. dBFS has expanded beyond traditional audio into other digital signal domains, including RF digitizers and analog-to-digital converter (ADC) testing. In RF systems, dBFS quantifies signal amplitudes relative to the digitizer's full-scale range, aiding in noise figure calculations and receiver sensitivity assessments.^[45] For ADC evaluation, test signals are typically applied at -1 dBFS to characterize dynamic performance without risking saturation, as detailed in industry application notes.^[46] ITU-R BS.1770 further incorporates dBFS for true-peak estimation in loudness algorithms, preventing inter-sample clipping in broadcast and streaming. Challenges in dBFS usage have arisen with advancements in bit depth and processing formats, prompting ongoing updates. The shift to 24-bit audio extends the theoretical dynamic range to about 144 dB below 0 dBFS, reducing quantization noise but requiring careful gain staging to exploit the full resolution. Floating-point formats, common in modern DAWs since the early 2000s, allow internal levels above 0 dBFS without hard clipping, facilitating non-destructive editing but necessitating export to fixed-point with headroom.^[47] More recently, industry practices have evolved toward integrated loudness metrics like LUFS—per EBU R128 (2010)—over pure peak dBFS, addressing listener fatigue from inconsistent volumes while retaining dBFS for peak compliance. Globally, AES and SMPTE guidelines have profoundly influenced professional audio workflows by promoting dBFS interoperability. AES Technical Document TD1004.1.15-10 recommends -16 to -18 LUFS with -1 dBTP (true peak) limits for streaming, standardizing mixing practices across studios.^[1] SMPTE RP155 (2006, revised) aligns analog reference levels to -20 dBFS in North American broadcast, ensuring seamless integration in post-production pipelines.^[48] These standards facilitate international collaboration in film, television, and music production. Key contributors included engineers like Tomlinson Holman, who in the early 1980s proposed refined metering approaches that integrated dBFS with monitoring standards, such as aligning -20 dBFS to 85 dB SPL for consistent leveling in digital workflows.^[49]

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[PDF] An Integrated Approach to Metering, Monitoring and Levelling
The European -18 dB reference is to a specific analog voltage level, and K-20 is referenced to a specific sound pressure level. The two systems can coexist ...
[44]
AES Standard » AES3-2009 (r2019) - Audio Engineering Society
AES3-2009 is a standard for serial transmission of two-channel digital audio data, including audio data semantics, channel status, and framing.Missing: dBFS key IEC 60908 EBU R68
[45]
https://www.ti.com/lit/pdf/slyt839
[46]
The History of the DAW - Yamaha Music
May 1, 2019 · The advent of the computer-based DAW in the early 1990s was the result of concurrent high-tech innovation and improvements in the areas of personal computers, ...
[47]
Audio Checkerboarding and Setting Audio Levels - Larry Jordan
May 15, 2011 · They told me that MP3 files are optimized for a maximum audio level of -6 dB, while AAC files are optimized for a maximum audio level of 0 dB.<|control11|><|separator|>
[48]
[PDF] How the ADC noise figure impacts RF receiver designs
This article explains how to calculate the noise figure of an RF-sampling ADC, and illustrate how the ADC noise figure affects RF signal- chain designs. Why the ...
[49]
AN-835: Understanding High Speed ADC Testing and Evaluation
AC or dynamic tests are typically made with the analog signal at the rated frequency with a signal power of 0.1 dB, 0.5 dB, or 1 dB below full scale (dBFS). If ...Missing: digitizers | Show results with:digitizers
[50]
-18 dBFS is the new 0 dBu - Blog - elysia.com
According to the EBU, the old analog “0 dBu” should now be equivalent to -18 dBFS (full scale). Digital consoles- and DAW users, therefore, hold fast: -18 dBFS ...
[51]
32-Bit Float Files Promise A Revolution In Field Recording
Sep 9, 2019 · These numbers are “fixed-point,” because they are whole numbers with no decimal point. Numeric values represent a discrete voltage level ...<|separator|>
[52]
[PDF] Video Standards - VideoQ
EBU R68, used in many countries, is mapping the alignment level of 0 dBu to -18 dBFS. SMPTE RP155 is mapping the US installations alignment level of +4 dBu to - ...Missing: key 60908