Let's talk about DSP audio processing

Aguilera

Let's talk about DSP audio processing [Copy link]

From compact discs (CDs) to super audio CDs (SACDs), DVD-Audio discs, and MP3 multimedia players, audio processing has made great progress as digital audio technology becomes more and more popular. This article focuses on audio processing with DSPs. Most audio processing is still done in the analog domain because early digital processing solutions based on general-purpose DSPs and external analog-to-digital converters (ADCs) and digital-to-analog converters (DACs) have significantly increased the additional cost of hardware and software programming. Therefore, it is difficult, time-consuming and costly to implement such solutions. Now there is a solution that integrates an audio-specific DSP and high-performance audio data converters on an integrated circuit. It can provide professional-quality digital sound processing with 112dB signal-to-noise ratio (SNR), full graphical user interface development and programming tools, and good price-performance ratio, allowing traditional analog systems to adopt digital technology with superior sound quality. The AD1954SigmaDSP is an example, which is a complete 26-bit, single-chip, 3-channel digital audio playback system with built-in DSP functions. Its main features include: 3 digital audio channels; a 7-band 48-bit stereo equalizer; a delay for loudspeaker position adjustment; a Phat Stereo spatial enhancement module; and a dual-band structured professional-quality dynamics processor. 1. Audio-specific DSP core A DSP core optimized for audio processing requirements. This user-configured DSP core has significant advantages over general-purpose DSP cores because it provides many features, such as hardware accelerators for double-precision filter calculations and dynamic processing. These features can significantly reduce the number of instruction cycles required for a given audio algorithm. This DSP core is based on a 26x22 multiply-accumulate engine with dual 48-bit accumulators. When the input word length is 24 bits, the internal resolution of the core is 26 bits in 3.23 format (3-bit exponent and 23-bit mantissa). Many audio algorithms require +12dB gain, and the additional 2 bits provide up to +12dB of gain, ensuring that no additional gain is required in most applications. All filters are calculated with 48-bit double-precision resolution using a dedicated hardware accelerator. Double-precision operation ensures that low-frequency infinite impulse response (IIR) filters work correctly and avoid finite-cycle problems that would otherwise produce artifacts. Graphical User Interface Graphical User Interface (GUI) makes it easy for experienced digital circuit engineers and analog circuit engineers who are familiar with their audio systems but do not want to get bogged down in low-level DSP programming of "bits and bytes" to add DSP to their systems. This tool not only allows intuitive operation, but also real-time control of the entire signal flow. It shows the signal flow graphically, so it is really intuitive to use. Designers can directly access and modify every parameter in the signal chain, including filter coefficients, volume settings, and dynamic processing functions, in real time. The GUI can be connected to the evaluation board through the printer port of the PC. In this way, any parameter changes can be sent through the SPI serial port and take effect immediately. 3. Design user-configured programs as needed Even if using internal programs is the simplest and most time-saving, design engineers may still want to configure special signal flows according to the needs of their system. The Graphic Compiler is a fully graphical program development tool. The graphical input tool allows system design engineers to draw user-configured signal flows and compile them into DSP programs at the press of a button. No line-by-line code development is required, making this workflow particularly friendly to analog circuit engineers. The DSP is used as a processing engine for 3-channel speakers. The processing process includes: total equalization module, 3-channel crossover module, single driver equalization module, each single driver dynamic range control module, delay module for driver position adjustment (phase correction), and DAC with 112dBSNR. 4. The importance of professional quality dynamic processing Small and medium-sized audio systems are often limited by the power of their amplifiers and speakers. In addition, due to the small size of the speaker, the frequency response of the low frequency speaker often has a premature natural roll-off at the lower frequencies. Therefore, it is very popular to use relatively strong equalization, especially in the bass region (bass boost circuit), in order to compensate for this imperfect setting of the sound. Finally, it is usually desired (if not required) that the system has a high maximum volume. The combination of the system's limited amplifier power, heavy bass equalization, and high overall system loudness will quickly saturate the amplifier and start to produce heavy distortion, which can be unsatisfactory or even annoying to the user. Several previous attempts to solve this problem have used simple clip detectors that can avoid clipping distortion, but the resulting artifacts are just as undesirable as clipping distortion. However, the use of the AD1954 SigmaDSP professional-quality, dual-band dynamics processor can control the limitations of this system without generating artifacts. 5. Improve the clarity and loudness of the system One is a transfer function without any dynamic processing, and the other is a transfer function with a compressor and limiter function with adjustable turning points. Due to the use of dynamic processing, the natural clipping level can be processed without distortion in the high volume area. This actually allows the user to increase the system volume by about 10dB. A 10dB increase in volume means that the sound pressure level has doubled, so the user can increase the loudness of the system to twice the original. This requires adjustments to the real-world factors that affect the sound. The transfer function with a user-configured DSP dynamic processor can be adjusted arbitrarily. It can combine several dynamic processing functions into a function curve. It has four typical functions, including: compression, limiting, expansion and noise threshold. Since this transfer function is fully programmable, these functions are very easy to implement and can be used individually or in combination. VI. Conclusion The introduction of audio-specific DSP technology has brought embedded audio system design into a new era. The processing performance, transformation technology and complex algorithms in the digital domain are all presented in graphical form, making their application more convenient and economical. This DSP technology enables design engineers to quickly develop or transplant their systems to the digital domain, thus fully utilizing the high quality of digital media.

alan000345

Audio seems to be difficult.

alan000345

The DSP microprocessor (chip) of DSP amplifier generally has the following main features: 1. One multiplication and one addition can be completed in one instruction cycle; 2. The program and data space are separated, and instructions and data can be accessed at the same time; 3. There is fast RAM on the chip, which can usually be accessed in two blocks at the same time through an independent data bus; 4. Hardware support for low-overhead or no-overhead loops and jumps; 5. Fast interrupt processing and hardware I/O support; 6. Multiple hardware address generators operating in a single cycle; 7. Multiple operations can be performed in parallel; 8. Support pipeline operation, so that operations such as instruction fetching, decoding and execution can be performed overlappingly. Compared with ordinary amplifiers, it is obviously much better. Ordinary amplifiers can only adjust: gain, high and low pass, and cannot be connected to a computer. DSP amplifiers can better manage amplifiers through computers. DSP amplifiers have several advantages that ordinary amplifiers do not have: 1. Integrating the DSP module into the amplifier saves wire cost and wire interference, and saves installation space in the car. 2. The power amplifier with DSP function can easily perform active frequency division, delay processing, and EQ adjustment, so that the complex environment of the car can be improved and the sound of the audio can be more pleasant and better to listen to!

电容器

DSP is indeed difficult to make.

风中客

I have a German Eton DSP board. The audio files cannot be restored after being saved. When I restart them, they return to the previous state. I would like to ask an expert for an answer.