Directory [hidden]
1 south latitude /PDIF
2 Number of sampling bits and sampling frequency
3 multi-tone numbers
4 Dynamic range
5 API interface
6 HRTF
7 ASIO
8 AC-3
9 DLS technology
10 SB 1394 standard
[edit edit]
S/PDIF is the abbreviation of Sony and Philips home digital audio interface, which can transmit PCM stream and surround sound compressed audio signals such as dolby digital and dts. Therefore, the most significant significance of adding S/PDIF function to the sound card is to make the computer sound card have more powerful equipment expansion ability. The performance of PDIF technology applied to sound cards is that sound cards provide S/PDIF input and S/PDIF output interfaces. If you have a digital decoder or speakers with digital audio decoding, you can use the S/PDIF interface as the digital audio output, and use an external DAC (digital-to-analog converter) for decoding to obtain better sound quality.
There are generally two kinds of S/PDIF interfaces, one is RCA coaxial interface, and the other is TOSLINK optical cable interface. RCA interface (non-standard) has the advantages of constant impedance and wide transmission bandwidth. In international standards, PDIF needs BNC interface for 75 ohm cable transmission. However, many manufacturers frequently use RCA interface or even 3.5mm stereo interface for S/PDIF transmission for various reasons.
On the multimedia sound card, S/PDIF is divided into two forms: output and input, commonly known as S/PDIF OUT and S/PDIF IN. The main function of the sound card's S/PDIF output is to transmit digital audio signals from the computer to various external devices. In the current mainstream products, the function of S/PDIF OUT has been very popular, and it is usually made on the main card or digital daughter card of sound card in the way of coaxial or optical fiber interface. The main function of S/PDIF in sound card is to receive PCM signals from other devices, and the most typical application is digital playback of CD records. Although all optical drives have the function of playing CD, the effect is different. The main reason is that the quality of DAC used in optical drive is different, which leads to different effects. However, if you have a two-pin S/PDIF IN socket on your sound card, you can connect to the audio digital Out interface of CD-ROM through a dual-core digital CD signal transmission line. In this way, when playing a CD record, the PCM signal on the CD is directly output to the sound card without DAC, and then the sound card performs D/A conversion or outputs it through S/PDIF OUT. The D/A conversion quality of general sound card codec chip is always better than that of DAC on CD-ROM, so the CD playback quality is effectively improved by S/PDIF technology.
[Edit] Sampling Quantity and Sampling Frequency
Audio signals are continuous analog signals, but computers can only process digital signals. Therefore, in order to process audio signals, computers must first perform analog-to-digital (A/D) conversion. This conversion process is actually the sampling and quantization process of audio signals, that is, converting analog signals that are continuous in time into digital signals that are discontinuous in time. As long as enough points are equidistant on the continuous quantity, the original continuous quantity can be simulated realistically. This process of "taking points" is called sampling. The higher the sampling accuracy (the more points are taken), the more realistic the digital sound will be. Among them, the sampling accuracy in the signal amplitude (voltage value) direction is called sampling resolution, and the sampling accuracy in the time direction is called sampling frequency.
The number of sampling bits refers to the amplitude of the audio signal represented by each sampling point. 8 bits can describe 256 states, while 16 bits can represent 65536 states. For the same signal amplitude, the quantization level of 16bit is more accurate than that of 8bit. This situation is like measuring in millimeters is more accurate than measuring in centimeters. Generally speaking, the higher the number of sampling bits, the clearer the sound.
Sampling frequency refers to the number of times that audio signals are sampled per second. The more sampling times per unit time, that is, the higher the sampling frequency, the closer the digital signal is to the original sound. As long as the sampling frequency reaches twice the highest frequency of the signal, the sampled signal can be accurately described. Generally speaking, the hearing range of human ears is between 20hz and 20Khz, so as long as the sampling frequency reaches 20Khz×2=40Khz, it can meet human requirements. At present, the sampling frequency of most sound cards has reached 44. 1 or 48Khz, which is the so-called CD sound quality level.
[Edit] Polyphony Numbers
In the naming of various sound cards, we often find the number 64, 128. Some users and even businesses mistakenly think that it is a 64-bit, 128-bit sound card, which represents the number of sampling bits. In fact, 64, 128 only represents the maximum polyphony that this card can achieve in MIDI synthesis. The so-called "polyphony" refers to the maximum number of sounds produced by MIDI music in one second. If the polyphonic value supported by the wave table is too small, some complicated MIDI music will be lost during synthesis, which will directly affect the playback effect. The more polyphony, the more realistic the sound effect, but this has nothing to do with the number of sampling bits. The current wave table sound card can provide polyphonic values above 128.
In addition, we should pay attention to the difference between "hardware supports polyphony" and "software supports polyphony". The so-called "hardware-supported polyphony" means that all polyphony numbers are generated by sound card chips, while "software-supported polyphony" is based on "hardware-supported polyphony" and increases polyphony numbers through software synthesis, but it needs CPU to drive it. At present, mainstream sound cards support hardware polyphony up to 64, while software polyphony up to 1024.
[Edit] Dynamic range
Dynamic range refers to the maximum range that the equipment owner can bear when the sound gain changes suddenly, that is, when the volume is suddenly or suddenly millimeter wave. The larger this value, the wider the dynamic range of the sound card, and the more it can show the mood and ups and downs of the work. Generally, the dynamic range of a sound card is about 85dB, and a sound card with a dynamic range over 90dB is a very good sound card.
= = Wave sound and MIDI music = =
The synthesis of waveform sound effect and MIDI music is the main function of sound card. Among them, the synthesis of wave sound effect is completed by ADC analog-to-digital converter and DAC digital-to-analog converter of sound card. Analog audio signals are converted into digital audio by ADC, and then stored in the form of files on disks and other media to become sound files. This kind of file is called waveform file, which is usually used. Wav extension, so it is also called wav file. Wave sound effects can realistically simulate various sound effects in nature. Unfortunately, wav files need to occupy a lot of storage space, and it is this shortcoming that has created the growth of MP3.
MIDI, the digital interface of musical instruments, is a communication standard for data exchange between computers and electronic musical instruments. MIDI files (usually with. Mid (as a file extension) records various control instructions for synthesizing MIDI music, including vocal instruments, channels used, volume and so on. Because MIDI file itself does not contain any digital audio signal, it occupies much less storage space than wav file. MIDI file playback needs to synthesize different sounds through the MIDI synthesizer of the sound card. There are two methods of synthesis: FM (frequency modulation) and Wave table (wave table).
Most cheap sound cards are synthesized by FM, and sine waves are generated by oscillators, and then superimposed into waveforms of various musical instruments. Because of the high cost of oscillators, even high-end FM synthesizers such as OPL3 only provide four oscillators and can only produce 20 kinds of polyphony, so the music sounds stiff and dull, and the synthesized color is obvious. Unlike FM synthesis, wavetable synthesis uses real sound samples for playback. Sound samples record waveform samples of various real musical instruments and store them in ROM or RAM on the sound card (to distinguish whether a sound card is a wave table sound card, just look at whether there is ROM or RAM memory on the card). At present, wave table synthesis technology is mostly used in middle and high-grade sound cards.
= = Output SNR = =
"Output signal-to-noise ratio" is an important factor to measure the sound quality of sound card. Its concept is the ratio of output signal voltage to noise voltage output at the same time, and the unit is decibel. The larger the value, the less noise mixed in the output signal and the purer the sound quality. Sound card, as the main output sound source of computer, requires relatively high signal-to-noise ratio. Because the sound output through the sound card needs to go through a series of complicated processing, there are many factors that determine the signal-to-noise ratio of the sound card Due to the serious electromagnetic radiation interference in the computer, it is difficult for the integrated sound card to achieve high signal-to-noise ratio. Generally, its signal-to-noise ratio is about 80dB. Generally, the signal-to-noise ratio of PCI sound cards is relatively high (most of them can easily reach 90dB), and some of them are as high as 195dB. A higher signal-to-noise ratio ensures a purer timbre when the sound is output, which can minimize noise. The quality of timbre depends on the sound card chip and the working condition of the sound card. If possible, you'd better listen to the sound card before buying it. If there is no audition, you can pay more attention to the evaluation of it by the surrounding media, which may be helpful for your purchase.
[edit] API interface
API means programming interface, which contains many instructions and specifications about sound localization and processing. Its performance will directly affect the expressive force of three-dimensional sound effects, mainly in the following aspects:
Direct sound 3D
Direct Sound 3D is a 3D effect positioning technology proposed by Microsoft, which is characterized by hardware independence. In the early sound card, many sound card chips did not have their own hardware 3D sound processing ability, so they all used this direct sound 3D to simulate stereo sound. The effects it produces are all produced by CPU through real-time operation, which consumes CPU resources. Therefore, all sound cards introduced since then have the so-called "hardware support DS3D" capability. If you listen to the manufacturer when you buy a sound card, don't believe that it is a good sound card. Its actual auditory effect depends on the strength of HRTF algorithm adopted by the sound card itself.
A3D
A3D is a patented technology developed by Aurel Company. It is developed on the basis of API interface of Direct Sound 3D. The biggest feature of "A3D" is that it can increase the realism of the new generation of game software interaction with 3D sound effects with accurate positioning, which is also commonly known as 3D positioning technology. There are currently three versions of A3D: 1.0, 2.0 and A3D3.0 The version of 1.0 includes two application fields of A3D surround and A3D interaction, with special emphasis on real sound field simulation in stereo hardware environment. In A3D 1.0, only 8 audio sources can be processed at the same time, and the sampling frequency is 22kHz. The AU8820 chip in AUREAL sound card adopts this technology. 2.0 is based on 1.0, adding acoustic tracking technology to further improve performance. A3D 2.0 can process 16 audio sources at the same time, and the sampling frequency reaches 48kHz. It is one of the best 3D audio technologies in today's positioning, and the AU8830 chip supports this technology. As for version 3.0, it has been proposed for a long time, but the future of A3D3.0 is still unknown because Aureal has been acquired by innovation.
Because Aureal's A3D technology has advantages in 3D positioning and interactive sound processing (which are two key parts), and supports direct sound 3D hardware acceleration, many game developers develop 3D games based on A3D. However, not every PCI sound card supports this technology because of its high implementation cost.
A3D surround
A3D surround sound absorbs the essence of A3D technology and surround sound decoding technology (such as Dolby's ProLogic and AC-3). Its outstanding feature is that only two ordinary speakers (or a pair of headphones) can accurately locate the sound in the surrounding three-dimensional space (that is, it can produce the same effect as five "virtual speakers"). Of course, these five audio streams don't need to be played with five actual speakers like the traditional "home theater", but are actually played with two speakers after A3D surround processing. This technology was awarded the "Virtual Dolby" certification by Dolby Lab.
EAX
EAX is an innovative company in its SB life! The full name of the standard proposed by series sound cards is ambient audio extension, that is, ambient sound effect. EAX is based on DS3D, but several unique sound commands are added to the latter. EAX focuses on rendering the changes and performances of various sounds under different environmental conditions, but its ability to locate sounds is not as good as A3D's. EAX suggests that users should be equipped with a 4-channel surround speaker system. At present, the main chips supporting EAX2 are EMU 10K 1 and MU 10K2, which is a famous innovation of SB Live! And Audigy series sound cards. At the same time, the chip also supports technologies such as A3D 1 and HRTF. It is a fine product in the currently popular compatible sound card.
Note: At present, the two major schools of API interfaces are A3D and EAX. When buying, it is best to find out what sound effects the selected sound card supports, what version it supports, whether it is software simulation or hardware support. These are very critical.
[edit] HRTF
Hrtf (Head Related Transfer Function) is the abbreviation of head related transfer function, which means "head corresponding transfer function" in Chinese, and it is also an important factor to realize three-dimensional sound effect. Simply put, HRTF is a kind of sound localization algorithm, and its practical function lies in deceiving our ears with digital sum algorithm, making us think that we are in a real sound environment. Three-dimensional positioning is realized by HRTF algorithm adopted by sound card chip, and the positioning effect is also determined by HRTF algorithm. Large companies like Aureal and Creative can develop powerful instruction set specifications and advanced HRTF algorithms and integrate them into their own chips. Of course, there are also some manufacturers who specialize in selling or formulating various HRTF algorithms for sound cards. The more famous ones are Sensaura 3D and Qsound. Sensaura 3D is provided by CRT Company. Sensaura supports most mainstream 3D audio APIs including A3D 1.0, EAX and DS3D, and is mainly used for sound card chips of ESS, Yamaha and CMI. Q3D developed by QSound mainly includes three parts, the first part is 3D sound effect and auditory environment model, the second part is stereo music enhancement, and the third part is virtual environment sound effect, which can provide environment simulation function similar to EAX, but the effect is still relatively simple, slightly inferior to Sensaura's huge and comprehensive performance index. In addition, C-MEDIA uses its own HRTF algorithm C3DX on CMI8738, which supports EAX and DS3D, and the actual effect is very general.
==IAS== IAS is the abbreviation of Interactive round-Sound. It is a patented audio technology developed by EAR(Extreme Audio Reality) company with the assistance of developers and hardware manufacturers. This technology can meet the needs of testing system hardware and managing all sound platforms. Developers only need to write a set of sound effect codes, and all audio hardware based on Windows 95/98/2000 will be supported through the same programming interface. IAS provides DS3D(Direct Sound 3D) support for sound designers and manages all sound resources. In addition, its sound output engine will automatically configure the best 3D audio solution, among which the four-channel sound card will be the primary target. The existing dual speaker platform can support DS3D.
[editor] ASIO
ASIO is the abbreviation of audio stream input and output, which can be translated as "audio stream input/output". Usually this is the performance that only professional sound cards or high-end audio workstations have. Using ASIO technology can reduce the delay of the system to the audio stream signal and enhance the audio processing ability of the sound card hardware. For the same sound card, if the delay time is 750 milliseconds when using MME driver, the delay time may be reduced to less than 40 milliseconds when using ASIO driver.
But not all sound cards support ASIO. ASIO not only defines the driving standard, but also requires the hardware support of the main chip of the sound card. Only those professional sound cards with high prices will consider the support for ASIO in their design. We often use sound cards, including the innovative past SB Live! This series belongs to the category of civil cards and is not equipped with ASIO drivers. However, the innovative SoundBlaster Audigy has begun to fully support ASIO technology.
Note: SB live! The main chip EMU 10K 1 itself supports ASIO, but this performance is not to innovate your own LiveWare! 3.0 drive. So when you let someone live! The audio processing software will report that ASIO has been found after replacing the driver with the driver of the E_mu APS recording card with the same specification! In addition, CMI8738 itself has the potential of ASIO, but so far there is no suitable driver to play it out.
[edit] AC-3
AC-3 is a completely digitized coded signal, so its official English name is "Dolby Digital", which was developed by the famous American Dolby Laboratory. Dolby's surround sound standard. AC-3 specifies six independent channels, namely, two front channels, two rear surround channels, a middle channel and a bass enhancement channel. Among them, the front, surround and center five channels are suggested to be full-band speakers, and the subwoofer is responsible for transmitting low-frequency and 80Hz subwoofers. The early AC-3 can only support 5. 1 channel at most. After continuous upgrading and improvement, the AC-3 6. 1 EX system has added the design of the rear surround center, allowing users to experience more accurate positioning.
At present, AC-3 is realized by hardware decoding and software decoding. Hardware decoding is to divide the sound room into 5. 1 channels through the decoder in the sound card that supports AC-3 signal transmission, and then output it through 5. 1 speakers. Software decoding means software decoding (for example, DVD playback software WinDVD and PowerDVD can support AC-3 decoding, and of course, the sound card must also support analog six-channel output. ), but one of the disadvantages of this working mode is that the decoding operation needs CPU to complete, which will increase the system burden, and the positioning ability of soft decoding is still inferior, and the sound field is scattered.
Although the software simulation of AC-3 has some defects, its cost is relatively low. At present, most middle and low-grade sound cards adopt this method.
[editor] DLS technology
The full name of DLS is "Down Loadable Sample", which means "downloadable sample sound library". Its principle is quite similar to that of the soft wave table, that is, the sound library is stored in the hard disk and transferred to the system memory when playing. But the difference is that after using DLS technology, MIDI synthesis does not use CPU to calculate, but depends on the audio processing chip of the sound card. The reason is that the data broadband of PCI sound card reaches 133Mb/ s, which greatly broadens the transmission channel between system memory and sound card. PCI sound card can use advanced DLS technology to store the tone of wave table in hard disk, and when MIDI is played, it is processed by sound card chip and transferred to memory. This not only saves the memory of the sound library needed by the traditional ISA wave table sound card, but also greatly reduces the CPU occupancy when playing MIDI. This not only provides a good MIDI synthesis effect, but also saves the memory of the sound library that must be equipped on the ISA wave table sound card. Moreover, this wave table library can be updated at any time and can be modified by DLS sound editing software, which is incomparable to traditional wave tables.
[Editor] SB 1394 standard
SB 1394 is an IEEE 1394 compatible standard proposed by innovative companies to realize high-speed digital audio transmission (about 400Mbps). The innovative SB 1394 standard ensures that the 1394 interface devices connected through SB 1394 can exert the maximum efficiency, and the transmission speed is as high as 400Mbps, which makes it possible to transmit large files between the host and peripherals at high speed. Sound blaster Audigy2 sound card has built-in SB 1394, which can be connected to external devices such as DV camera through IEEE 1394 standard interface, and can be connected to 63 computers to play online games with low delay.