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There are 50 adjustment parameters for DSP audio. Once you know how to use them, you're really amazing!
Date:November 7, 2025    Views:84

    There are many parameters related to adjusting audio with DSP (Digital Signal Processing), covering multiple dimensions of signal processing. The following are the main parameter classifications and specific contents:
I、Basic volume and gain control
    1- Input Gain: Adjust the amplitude of the input signal to avoid overload or weak signal.
    2- Output Gain: Adjust the output amplitude of the signal processed by the DSP to match the back-end equipment (such as power amplifiers).
    3- Volume: The overall loudness control of the signal, similar to the output gain function in some scenarios.
II、Frequency equalization (EQ) parameter
    4- Center Frequency: The purpose of EQ regulation
    5- Mark frequency points (such as 100Hz, 1kHz), and enhance or attenuate for specific frequency bands.
    6- Gain: Increase (+dB) or attenuate (-dB) the signal amplitude of the center frequency and nearby frequency bands.
    7- Bandwidth/Q Value (Bandwidth/Q) : Controls the frequency range affected by EQ regulation. The higher the Q value, the narrower the affected range (precise regulation). The lower the Q value, the wider the range (large range adjustment).
    8- Filter Type:
    9- High-pass Filter (HPF) : Filters out signals below the set frequency (such as removing low-frequency noise).
    10-Low-pass filter (LPF) : Filters out signals higher than the set frequency (such as limiting high-frequency distortion).
    11- Bandpass Filter (BPF) : Only retains signals in specific frequency bands.
    12- Band-stop Filter (BEF/ notch) : Attenuates signals in specific frequency bands (such as eliminating howling frequencies).
    13- Parametric EQ: A flexible filter that can independently adjust the center frequency, gain, and Q value.
    14- Graphic EQ: A slider adjustment at fixed frequency points, which is intuitive but less flexible.
III、Dynamic processing parameters
    15- Compressor Parameters:
    16. Threshold: Compression begins when the signal exceeds this level (e.g., -10 DBFS).
    17+ Ratio: The compression ratio when the input signal exceeds the threshold (for example, 4:1, if the input exceeds the threshold by 4dB, the output only exceeds by 1dB).
    18-attack Time (Attack Time) : The response time for the compressor to start working after the signal exceeds the threshold (e.g., 10ms).
    19- Release Time: The recovery time when the compressor stops working after the signal drops below the threshold (e.g., 100ms).
    20- Make up Gain: Compensates for the loudness loss of the compressed signal.
Limiter parameters:
    21- Threshold: The maximum level that the signal is not allowed to exceed (e.g., 0dBFS, to prevent overload).
    22- Ratio (Ratio) : Usually ∞:1 (all signals exceeding the threshold are restricted).
    23-attack/release time: The same as the compressor, with a greater emphasis on rapid response to avoid clipping.
    24- Expander parameters:
    25- Threshold: The signal begins to attenuate below this level (to reduce background noise).
    26- Ratio: The attenuation ratio of the input signal when it is lower than the threshold (for example, 1:2, input lower by 2dB, output lower by 4dB).
   27-attack/release time: Controls the response speed of the extender.
    28- Noise Gate parameters:
    29- Threshold: Signals below this level are completely muted (low-level noise is completely eliminated).
    30- Open Time: The delay time for the noise gate to open after the signal exceeds the threshold.
    31- Close Time: The delay time for the noise gate to close after the signal drops below the threshold.
Ⅳ、Time and space processing parameters
    32- Delay parameter:
    33- Delay Time: The duration of signal delay output (e.g. 50ms, used to create a sense of space or synchronous multi-channel).
    34- Feedback: The repeated attenuation ratio of the delayed signal (such as 30%, controlling the number of echoes).
    35- Mix ratio: The volume ratio of the original signal to the delayed signal.
    36- Reverb parameters:
    37- Pre-delay: The time interval between direct sound and reverberation sound (e.g., 20ms, simulating room size).
    38- Decay Time: The time it takes for the reverberation sound to decay from its peak to its disappearance (for example, 1.5 seconds, which determines the sense of spaciousness in the space).
    39- Early Reflections: The intensity and duration of the first sound rebound (simulating room boundary reflections).
    40- Wet/Dry ratio: The ratio of the reverberation signal to the original signal.
Channel processing
    41- Channel Balance: The volume ratio of the left and right channels.
    42- Sound Image (Pan) : The positioning of the signal in the stereo field (e.g., 30% left, 70% right). 
Ⅳ、Distortion and special effect parameters
    43- Total Harmonic Distortion (THD) : Intentionally introduced harmonic components (such as guitar distortion effects).
    44- Modulation Effect:
    Tremolo: The frequency and depth of periodic changes in volume.
    45- Flanger: Rapid phase difference modulation between the delayed signal and the original signal.
    46- Phase shifting (Phaser) : The phase shift effect produced when a signal passes through multiple bandpass filters.
Ⅴ、Other key parameters
    47- Sample Rate: The sampling frequency of the signal processed by the DSP (such as 44.1kHz, 48kHz, which affects the upper limit of the frequency response).
    48-bit Depth: Signal quantization accuracy (such as 16-bit, 24-bit, affecting dynamic range and noise level).
    49- Phase: Inversion of signal phase (such as polarity inversion, used to correct multi-device phase cancellation).
    50- Crossover Frequency: The separation point of high and low frequency signals in a crossover (such as 2kHz, used for crossover in multi-speaker systems).
    These parameters are combined and used according to specific application scenarios (such as music production, audio system debugging, voice processing, etc.), and the precise control and optimization of audio signals are achieved through DSP algorithms.



  

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