In-depth understanding of digital audio processing technology

Digital audio processing technology is a technology that uses digital signal processing algorithms to operate audio signals.

First, the rationale

Sampling and quantization

The audio signal is an analog signal, and digital audio processing should first be sampled, that is, the amplitude of the analog audio signal should be taken according to a certain time interval. For example, a common sample rate is 44.1 kHz (the standard for CD audio), which means that the audio signal is amplitude taken 44,100 times per second.

Quantization converts the sampled amplitude value into a numeric encoding. The number of quantization bits determines the dynamic range of the sound, for example, 16-bit quantization can provide a large dynamic range and can better restore the details of the sound.

Encoding and compression

Encoding is the organization of quantized digital audio data in a certain format. For example, PCM (Pulse Code Modulation) is a basic way of coding.

Compression is to reduce the storage space and transmission bandwidth of audio data. Common lossy compression formats such as MP3, which reduces file size by removing audio information that is less sensitive to the human ear; Lossless compression formats, such as FLAC, reduce file size without losing audio quality.

Second, the main processing technology

filtering

Low-pass filtering: Allows low-frequency signals to pass through, blocking high-frequency signals. It can be used to remove high-frequency noise from the audio, making the sound smoother.

High-pass filtering: In contrast to low-pass filtering, high-frequency signals are allowed to pass through and low-frequency signals are filtered out. In some audio processing scenarios, it can be used to highlight high-frequency details in sound.

Bandpass filtering: Allows only signals in a certain frequency range to pass through, and can be used to extract or remove sound in a certain frequency band.

Equalization (EQ)

Adjust the amplitude of the audio signal on different frequencies. For example, boosting the low frequencies makes the sound sound thicker, and boosting the high frequencies makes the sound brighter and clearer. There are different types of EQ, which adjusts the gain of preset bands through an intuitive graphical interface, and parametric EQ, which allows you to set parameters such as center frequency, bandwidth, and gain more precisely.

reverberation

Simulate the reflection of sound in different spatial environments. By adding reverberation, you can make the sound sound like it's coming from a different space, such as a cathedral, a concert hall, or a small room. The parameters of reverberation include reverberation time, pre-delay, etc., the reverberation time determines the length of time after the sound is reflected, and the pre-delay refers to the time interval between the direct sound and the first reflected sound.

Compression and limiting

Compression: Reduces the part of the audio signal that has a large dynamic range. When the input signal exceeds a set threshold, the compressor reduces the gain of the signal by a certain percentage. This avoids high peaks in the audio signal and makes the sound more balanced and stable overall.

Clipping: Similar to compression, but limiting is when the signal exceeds a threshold, limiting the maximum amplitude of the signal to a fixed value to prevent excessive clipping distortion of the signal.

Delay and echo effects

Delay refers to the copying of an audio signal at certain intervals and delaying playback. By adjusting the delay time and feedback parameters, an echo effect can be created. In music production, it is often used to create a sense of space or special sound effects.

Third, the field of application

Music production

Digital audio processing is ubiquitous in the recording, mixing, and mastering process. These techniques are indispensable, from the modification of individual instruments and vocals, to the timbre shaping, spatial creation and dynamic balance of the entire musical composition.

Audio broadcasts

Radio stations need to process audio signals, such as adjusting volume, equalization, compression, etc., to ensure the quality of broadcast sound, and also encoding and compressing to meet the requirements of broadcast transmission.

Film and television production

In film and television production, digital audio processing is used to process dialogue, music, and sound effects in films. For example, add appropriate reverb to dialogue to match the context of the scene, filter and equalize sound effects to enhance realism, and more.

Virtual Reality (VR) vs. Augmented Reality (AR)

To create realistic virtual environments and augmented reality experiences, special processing of audio is required. For example, adjust the direction, distance, and reverberation effect of sound in real time based on the user's position in the virtual space.

Fourth, the development trend

Applications of artificial intelligence and machine learning

Leverage AI algorithms for audio analysis, classification, and processing. For example, automatically identify the type of instrument in the audio and make targeted timbre optimizations, or automatically adjust the rhythm and timbre of the music based on the user's emotional state.

High-resolution audio

As technology evolves, audio processing with high sampling rates (e.g., 192kHz or beyond) and high quantization bits (e.g., 24-bit or higher) becomes a trend to deliver a high-fidelity audio experience that is closer to the original sound.

Real-time processing power increases

In scenarios such as live streaming and interactive music creation, the real-time requirements for digital audio processing are getting higher and higher. Future technologies will continue to increase the speed and efficiency of real-time processing and reduce latency.