Construction and Research of Digital Sound Effects Based on LabVIEW and Speedy-33
2026-04-06 06:07:12··#1
Products: LabVIEW, Academic Products The Challenge: The implementation and analysis of various sound effect algorithms (such as virtual surround sound, reverb, equalization, etc.) are essential topics in university courses such as digital audio signal processing. If students only engage in theoretical analysis without building various sound effect units on a hardware platform and then actually listening to them, they lack intuitive understanding, resulting in unsatisfactory teaching effectiveness. Implementing these on a general DSP development board using C, assembly, or other software programming is extremely labor-intensive, and the sound effect unit parameters cannot be adjusted in real-time during program execution, making quantitative analysis and comparison inconvenient. Therefore, a development platform that can quickly and flexibly build various sound effect algorithms is a major challenge. The Solution: Various sound effect units are quickly built using LabVIEW graphical programming. The sound effect algorithm programs are then downloaded to the NI Speedy-33 DSP hardware development module for implementation. Finally, the parameters are adjusted in real-time during program execution to complete the analysis and testing of the sound effect units. I. Introduction To improve sound playback quality and achieve various sound effects, various audio devices we use today (such as home stereo systems, MP3 players, MD players, etc.) incorporate various sound effects processors through software programming or hardware modules. This article describes how to build several commonly used sound effects processors on the SPEEDY-33 DSP hardware development module, adjust the relevant parameters of the sound effects processors, test their effects, and provide a detailed analysis. II. Sound Effects Processor Development Platform and Testing Environment 1. Sound Effects Processor Development Platform: The NI SPEEDY-33 is actually a DSP development board. The CPU uses the TI TMS320VC33-150 chip, which is a 32-bit floating-point processor with a processing capacity of 150 MFLOPS (75 MIPS), which is fast enough for implementing general sound effect algorithms. The development board comes with A/D and D/A modules, both of which can process dual-channel signals simultaneously, with a quantization precision of 16 bits and a maximum conversion speed of 48 KSPS. Therefore, it is evident that the SPEEDY-33 is very suitable for processing stereo audio signals, and its sound quality can theoretically reach CD quality levels. Various sound effect algorithms are implemented using LabVIEW programming and then downloaded to the SPEEDY-33 for execution. The parameters of the sound effect processor are then adjusted in real time through the LabVIEW program's front panel, and the changes in the time and frequency domain waveforms of the audio signal are observed. Simultaneously, the changes in sound are heard using headphones or speakers. When selecting a sound effect processor development platform, the traditional development model was also considered: developing various sound effect processors using C and assembly language on a DSK board (such as the TI C6713 DSK). However, compared to the model adopted in this paper, the programming workload is large, the development cycle is long, and the human-computer interaction is poor. 2. Test Environment: I converted the stereo analog signal output from the CD player into a digital signal (sampling rate: 48kHz, precision: 16-bit) using the SPEEDY-33's A/D module. After digital sound effect processing by the DSP, the signal was output by the D/A module, and then amplified before being played through speakers or headphones. The amplifier unit used is a YAMAHA RX-350, rated power 90W/channel. During testing, a pass-through mode was used, meaning the audio signal received no further processing by the amplifier unit. The speakers used are PIONEER CS-222Z, maximum power 60W/channel, 2-way design. In the test environment, the left and right speakers were spaced 1.8 meters apart, and the listening position was 2 meters directly between the two speakers, centered on each other. SONY MDR-V300 semi-closed stereo headphones were used. See Figure 1 for the entire test platform and environment: [align=center] Figure 1 Test Platform and Environment[/align] III. Construction and Testing of Sound Effects Units 1. Virtual Surround Effector: Virtual surround is a technology often referred to as non-standard surround sound technology. Non-standard surround sound systems, based on two-channel stereo, do not add channels or speakers. They process the sound field signal through circuitry before playback, making the listener feel that the sound is coming from multiple directions, creating a simulated stereo sound field. Various algorithms can be used to achieve non-standard surround sound; this example uses the widely used SRS algorithm. SRS (Sound Retrieval System) does not focus on hardware to create a three-dimensional sound field, but rather on auditory psychology, simulating a three-dimensional sound field to make the listener feel immersed in it. In reality, this "three-dimensional sound field" does not exist; it is merely an illusion, just like 3D movies and 3D pictures, which use technology to transform two-dimensional plane objects into three-dimensional spatial images. SRS psychologically and subjectively restores the sound wave state created by the original sound source at both ears, reproducing the location and spatial distribution of the original sound source, giving the listener a sense of immersion. (1) The system uses the Modified SRS algorithm, and its implementation block diagram is shown in Figure 2: [align=center] Figure 2 Modified SRS algorithm block diagram[/align] (2) Test results and analysis: The units that work in the effect unit are the LR and L+R units. Let's first look at the time domain and frequency domain waveforms of the same music clip after processing by LR and L+R. [align=center] Figure 3 Time domain and frequency domain waveforms of LR and L+R signals[/align] As can be seen from Figure 3, the L+R signal has a larger amplitude and energy than the LR signal; the energy of the LR signal is more dispersed in the spectrum, while the energy of the L+R signal is relatively more concentrated in the low and mid frequency bands, which is consistent with the actual listening results. (3) Conclusion: Using the SRS virtual surround sound effect unit in the audio system can make the sound field wider and enhance the sense of space; at the same time, it increases the sense of immersion and presence of the music - making us feel as if we are in a concert hall or other performance venue. However, SRS also has negative effects: it will worsen the sense of location and layering of music (such as the location and layering relationship of various instruments and voices in symphonic and choral music); in addition, virtual surround sound has high requirements for listening position, so applying SRS virtual surround technology to headphones is a good choice. 2. Reverberation: There are many ways to implement a reverberation, but its basic unit is to delay the sound for a period of time and then superimpose it with the direct sound. The following is a relatively simple reverberation algorithm. (1) Reverberation algorithm block diagram: [align=center] Figure 4 Reverberation algorithm block diagram[/align] (2) Test results and analysis: [align=center] Figure 5 Original signal and signal spectrum after adding reverberation[/align] Figure 5 shows the spectrum of the original signal of the same music clip (left) and the spectrum after adding reverberation (right). From Figure 5, we can clearly see that the spectrum of the signal becomes richer after adding reverberation, and the low frequency part is enhanced. This is consistent with our actual listening results. That is, after adding reverb, the sound becomes "fuller" and "smoother", and the low frequency part becomes "thicker". However, if the coefficient of the effect is not chosen properly, such as the coefficient b being too large (too much delay sound component), the sound will become muffled and details will be lost. (3) Conclusion: Reverb plays an important role in sound effects. Adding some reverb appropriately will modify the sound and make it sound more pleasant. For this reason, the production staff will add a certain amount of reverb to the various music CDs and tapes we buy during the early recording process. At the same time, the playback equipment such as speakers and the walls and ceilings in the listening environment will add some reverb to the played sound signal. Therefore, our home audio playback equipment generally does not have a separate reverb effect (except for karaoke equipment). 3. 5-Band Graphic Equalizer: Equalizers are commonly used sound effects in audio systems. Some systems use independent graphic equalizer modules, while some handheld devices (such as Walkmans and MP3 players) use equalizers with multiple sound effect selection switches such as "Jazz", "Pop", and "Rock". However, they are essentially adjustable gain filter banks. (1) Equalizer principle: [align=center] Figure 6 Algorithm model of 5-band equalizer[/align] H1(n) is a low-pass filter, H2(n), H3(n), and H4(n) are three band-pass filters, and H5(n) is a high-pass filter. By adjusting their respective gain coefficients G1 to G5, we can adjust the sound of the corresponding frequency band and obtain the desired effect. See Figure 7 for the amplitude-frequency characteristic curve of the equalizer: [align=center] Figure 7 Amplitude-frequency characteristic curve of 5-band equalizer[/align] (2) Conclusion: Using an equalizer can significantly change the sound effect, such as increasing the bass or treble components of music. However, this change doesn't achieve the same realistic and natural effect as improving the performance of playback devices (speakers, headphones). Furthermore, improper use of digital filters or their parameters can introduce phase distortion or harmonic distortion in the audio signal, both of which affect sound quality. 4. Comprehensive Application: By encapsulating the various sound effect programs mentioned above into individual SubVIs, and adding volume and balance modules, a comprehensive sound effect adjuster can be built on the SPEEDY-33 platform. This allows us to test the effects of using multiple sound effects together. We can also enhance the program's front panel to make it look more like a "sound system." See Figure 8: [align=center] Figure 8 Comprehensive Sound Effect Program Front Panel[/align] IV. Conclusion Using LabVIEW graphical programming software and SPEEDY-33 hardware modules, we quickly implemented various sound effect algorithms and built various sound effect units. This is much easier and less labor-intensive than using programming languages such as C and assembly. Using this system, students can focus their main efforts on algorithm research, thus enabling them to build more complex and higher-performing sound effects devices with the same time and effort. Furthermore, LabVIEW software boasts strong human-computer interaction capabilities; we don't need to write additional host computer programs, and can build a beautiful and practical interactive interface using the front panel controls. Through this interface, we can change various parameters in real time during program execution to achieve different effects for comparison, and also easily observe and analyze the time-domain and frequency-domain waveforms of audio signals. In conclusion, this system is a highly flexible, convenient, and powerful experimental research platform for digital audio signal processing. Due to space limitations, this article has been abridged.