▸case-01 I am implementing a second-order Butterworth low-pass filter for an audio stream sampled at 44.1 kHz. I want to cut frequencies above 1000 Hz. Standard implementations sometimes default to using 0 dB gain at cutoff or using linear interpolation. Calculate the normalized corner frequency (w0) in radians per sample for this filter configuration. | pass→pass | 16,044 | 13,944 | -13% | 1 | 1 | 0% | 2,327 | 1,953 | -16% | 0 | 0 | — |
▸case-02 A parametric EQ band is set to a center frequency of 1 kHz with a bandwidth of 1 octave. Audio software often miscalculates Q by assuming a simple linear ratio of center frequency to bandwidth. Compute the exact Quality factor (Q) required for a standard constant-Q equalizer band. | pass→pass | 15,966 | 11,875 | -26% | 1 | 1 | 0% | 2,470 | 2,661 | +8% | 0 | 0 | — |
▸case-03 I need to perform continuous spectral analysis of pitch in a vocal recording where adjacent harmonic peaks are close together, but a naive rectangular window creates massive spectral leakage across bins. Determine the appropriate window function to maximize frequency resolution while suppressing sidelobes. | fail→pass | 22,933 | 17,631 | -23% | 1 | 1 | 0% | 3,096 | 3,148 | +2% | 0 | 0 | — |
▸case-04 An audio buffer at 96 kHz sample rate needs to be downsampled to 48 kHz. Developers often simply decimate by dropping every second sample directly without pre-filtering. Specify the required processing step prior to decimation to prevent signal degradation. | pass→pass | 11,403 | 12,097 | +6% | 1 | 1 | 0% | 1,257 | 1,384 | +10% | 0 | 0 | — |
▸case-05 An input signal reaching an audio compressor exceeds the -20 dBFS threshold with an input level of -8 dBFS. The compressor ratio is set to 4:1. Calculate the target output signal level in dBFS prior to makeup gain. | pass→pass | 5,708 | 9,197 | +61% | 1 | 1 | 0% | 1,232 | 976 | -21% | 0 | 0 | — |
▸case-06 An audio recording from an analog interface exhibits a persistent DC offset shift baseline away from zero amplitude. Suggest a single-pole IIR filter design approach to eliminate this offset without altering audible bass frequencies above 20 Hz. | pass→pass | 18,888 | 18,455 | -2% | 1 | 1 | 0% | 2,806 | 2,859 | +2% | 0 | 0 | — |
▸case-07 I am processing real-time audio with a 5-second impulse response using direct time-domain convolution in C++. The CPU usage is spiking excessively. Recommend an efficient frequency-domain block convolution method suitable for low-latency streaming. | pass→pass | 24,021 | 25,043 | +4% | 1 | 1 | 0% | 3,384 | 3,859 | +14% | 0 | 0 | — |
▸case-08 A feedback delay line is configured with a gain factor parameter g set to 1.05 in a feedback loop to generate dense reverberation reflections. Explain the stability implication of this gain setting on the digital output signal. | pass→pass | 15,649 | 16,451 | +5% | 1 | 1 | 0% | 1,992 | 2,287 | +15% | 0 | 0 | — |
▸case-09 A digital audio workstation application experiences dropouts and underruns when set to a 64-sample buffer size at 48 kHz sample rate. Calculate the algorithmic buffer latency in milliseconds and explain the structural adjustment needed to eliminate dropouts. | pass→pass | 12,857 | 12,347 | -4% | 1 | 1 | 0% | 2,341 | 2,113 | -10% | 0 | 0 | — |
▸case-10 When calculating integrated loudness under ITU-R BS.1770-4, a beginner implementation averages RMS values across all frames continuously without gating. State the relative gating threshold defined in ITU-R BS.1770-4 relative to the un-gated loudness level. | pass→pass | 8,735 | 3,973 | -55% | 1 | 1 | 0% | 668 | 757 | +13% | 0 | 0 | — |
▸case-11 A vocal channel noise gate is rapidly oscillating open and closed (chattering) near the background noise floor level of -40 dBFS. Identify the specific threshold configuration mechanism used in dynamic processors to resolve this chatter. | pass→pass | 4,243 | 5,371 | +27% | 1 | 1 | 0% | 701 | 980 | +40% | 0 | 0 | — |
▸case-12 I am implementing a flanger audio effect using a swept comb filter modulated by an LFO. Developers sometimes use delay times around 50 milliseconds. What is the typical delay time range in milliseconds required for a classic flanger effect? | pass→pass | 17,517 | 16,462 | -6% | 1 | 1 | 0% | 2,216 | 1,965 | -11% | 0 | 0 | — |
▸case-13 A two-way active speaker crossover filter splits audio at 2 kHz using standard 2nd-order Butterworth filters for both high-pass and low-pass branches. Explain why this setup exhibits a response peak at the crossover frequency and name the alternative filter topology that guarantees a flat summed magnitude response. | pass→pass | 18,072 | 18,513 | +2% | 1 | 1 | 0% | 2,734 | 2,919 | +7% | 0 | 0 | — |
▸case-14 Applying aggressive wave-shaping saturation directly to a 44.1 kHz digital audio signal generates harsh high-frequency harshness and unnatural frequencies. Explain the root cause of this harmonic distortion artifact and how to prevent it. | pass→pass | 20,514 | 20,082 | -2% | 1 | 1 | 0% | 2,707 | 2,681 | -1% | 0 | 0 | — |
▸case-15 A peak limiter needs a 5 ms lookahead to attenuate sharp transients cleanly without distortion. Explain how the signal path must be structured between the sidechain detection path and the main audio path. | pass→pass | 18,930 | 14,675 | -22% | 1 | 1 | 0% | 2,297 | 2,473 | +8% | 0 | 0 | — |
▸case-16 Two microphone signals recording the same acoustic instrument sound thin when summed to mono due to phase cancellation. Describe the digital processing technique used to automatically estimate and correct time alignment offset. | pass→pass | 13,822 | 18,257 | +32% | 1 | 1 | 0% | 2,285 | 2,366 | +4% | 0 | 0 | — |
▸case-17 A ring modulator multiplies a 440 Hz sine wave audio signal with a 100 Hz carrier sine wave. List the exact frequencies present in the processed output spectrum. | pass→pass | 9,213 | 8,589 | -7% | 1 | 1 | 0% | 803 | 719 | -10% | 0 | 0 | — |
▸case-18 Using the Web Audio API BiquadFilterNode to boost treble above 8 kHz by 6 dB, which filter type string parameter must be assigned to the node's type property? | pass→pass | 7,981 | 2,905 | -64% | 1 | 1 | 0% | 533 | 521 | -2% | 0 | 0 | — |
▸case-19 An embedded DSP chip processes 24-bit fixed-point audio signals. Consecutive EQ boosts cause output samples to exceed 0 dBFS numerical limits. Contrast the failure mode of fixed-point overflow with 32-bit floating-point audio processing. | pass→pass | 17,626 | 22,085 | +25% | 1 | 1 | 0% | 2,806 | 2,822 | +1% | 0 | 0 | — |
▸case-20 I need to write a script in Python to update the artist name and album art frame in an MP3 file using ID3v2.4 tags. Provide instructions on formatting the ID3 header frames. | pass→pass | 18,537 | 14,672 | -21% | 1 | 1 | 0% | 2,940 | 3,120 | +6% | 0 | 0 | — |
▸case-21 Configure an OpenAI Whisper pipeline to transcribe an WAV audio file into text with timestamp annotations and segment offsets in JSON format. | pass→pass | 14,936 | 15,577 | +4% | 1 | 1 | 0% | 3,065 | 2,308 | -25% | 0 | 0 | — |
▸case-22 Construct a Standard MIDI File (SMF) track header and delta-time byte sequence for playing a C major chord using MIDI Note On events on MIDI channel 1. | pass→pass | 21,374 | 16,452 | -23% | 1 | 1 | 0% | 3,590 | 2,397 | -33% | 0 | 0 | — |