0 Preface
Digital audio processing refers to the codec processing technology for audio to reproduce the realistic effect of sound. It is the key technology of broadband network multimedia and mobile multimedia communication. Audio Codec'97 (audio digital signal encoder / decoder) is one of them A technical standard for sound recording and playback, referred to as AC′97. AC′97 adopts a dual integrated structure, namely Digital Controller (Digital Signal Controller) and Audio Codec (audio codec), which makes the analog-to-digital converter ADC and digital / Analog converter DAC conversion module is independent, as far as possible to reduce the impact of EMI (electromagnetic interference).
Using FPGA, you can achieve complex logic control and parallel processing of a large amount of audio data. FPGA provides a programmable clock generator to meet the requirements of audio and video processing, wide clock range, small phase jitter (Phase Jitter), and provide the system with Controllable delay.
1 AC-Link audio encoding / decoding principle
AC-Link is a 5-wire serial time division multiplexed I / O interface connecting Digital Controller and Audio Codec. The fixed clock frequency of 48kHz is derived from the serial bit clock 12.288MHz divided by 256. It supports one controller and up to 4 codes. AC-Link can only transmit PCM (Pulse Code Modulation) signals with a fixed sampling rate of 48kHz, and the word length is from 16Bit to 20Bit. PCM signals with other sampling rates must be converted to 48kHz by SRC (Sampling Rate Conversion).
The timing of the AC-Link interface is shown in Figure 1.The input and output audio data and the read and write commands of the control register are organized in a frame.An input or output is divided into 12 time slots, and each time slot has a sampling resolution of 20 bits. The controller divides the 12.288MHz clock by 256 to generate a SYNC signal, which is used to mark the beginning of an input (output) frame.
Figure 1 Bidirectional AC-Link data frame and time slot allocation
As can be seen from Figure 1, in addition to 12 20-bit data / command (data / status) multiplexing time slots for each input (output) frame, there is also a special 16-bit first frame time slot. Used to mark whether this frame is available, if this frame is available, then the corresponding time slot in this frame is valid data.
As shown in Figure 2, PCM converts continuously changing analog signals into digital encoding through three steps of sampling, quantization, and encoding. PCM encoding is the highest fidelity level encoding with good sound quality but large volume. AC-Link can transmit 48KHz fixed samples The rate of PCM signal can be from 16Bit to 20Bit. The PCM signal of other sampling rate must first be converted to 48KHz through SRC (Sample Rate Conversion).
Figure 2 AC-Link audio encoding and decoding process
If the word length of the PCM signal is lower than that of the DAC, the Controller will automatically shift the PCM signal to align its MSB (Most Significant Bit), and the low bit will be filled with 0. If the word length of the PCM signal is higher than that of the DAC signal , Then you must first reduce the word length through Dither (dithering) or directly transfer it to Codec through the AC-Link interface. The LSBs (Least Significant Bit) are removed. The output of the DAC is a stepped or pulsed signal, which must be filtered and shaped by LPF (Low Pass Filter) to restore the original audio signal. funcTIon ImgZoom (Id) // Re-set the picture size to prevent the form from being broken {var w = $ (Id) .width; var m = 650; if (w <m) {return;} else {var h = $ (Id) .height; $ (Id) .height = parseInt (h * m / w); $ (Id) .width = m;}} window.onload = funcTIon () {var Imgs = $ ("content"). getElementsByTagName ( "img"); var i = 0; for (; i
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