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Digital audio commonly used audio protocols are PCM, I2S, AC97. PCM is mainly used for voice communication, and the general sampling frequency is 8K. I2S is a protocol between chips, mainly between AP and codec, 3 lines commonly used by I2S.
MCLK: It is called the master clock, also called the system clock (Sys Clock), which is 256 times, 384 times or 512 times the sampling frequency.
BCLK: Serial clock SCLK, also called bit clock.
LRCK: Frame clock, used to switch the left and right channel data. LRCK of “1” means that the data of the left channel is being transmitted, and “0” means that the data of the right channel is being transmitted. The frequency of LRCK is equal to the sampling frequency.
MCLK is used for the system, SMCLK is used for high-speed peripherals, and ACLK is mainly used for low-speed peripherals.
AC97 is mostly used for PC, it contains control signal itself, unlike I2S which requires additional control signal such as I2C.
Generally speaking, the output of the encoder of audio codec (such as UDA1341)/input of the decoder is linear PCM format.
Pcm, usually a voice signal. I2s is generally an audio signal used between AP and codec AC97 is used for AP and codec AC97 has its own control signal, while I2S requires additional control signals such as I2C.
AC’97 adopts a dual integrated structure, namely Digital Controller (digital signal controller) and Audio Codec (audio codec), which makes the analog/digital converter ADC and digital-to-analog converter DAC conversion modules independent, and minimizes 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 a wide range of clocks and low phase jitter (Phase Jitter) required by audio and video processing, and provide the system with Controllable delay.
AC-Link is a 5-wire serial time-division multiplex I/O interface that connects Digital Controller and Audio Codec. The fixed clock frequency is 48kHz and is divided by 256 from the serial bit clock 12.288MHz. 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. The word length ranges from 16Bit to 20Bit. PCM signals with other sampling rates must be converted to 48kHz by SRC (Sampling Rate Conversion).
AC-Link interface timing input and output audio data and control register read and write commands are organized in one frame, an input or output is divided into 12 time slots, and each time slot has a 20-bit sampling resolution. The controller uses a 12.288MHz clock Divide by 256 to generate a SYNC signal, which is used to mark the beginning of an input (output) frame.
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 frame first time slot, which is mainly used to mark this frame Whether it is available, if this frame is available, then the corresponding time slot in this frame is valid data.
PCM converts continuously changing analog signals into digital codes through the three steps of sampling, quantization, and coding. PCM coding is the highest fidelity level coding with good sound quality but large volume. AC-Link can transmit PCM signals with a fixed sampling rate of 48KHz. The length can be from 16Bit to 20Bit. PCM signals with other sampling rates must be converted to 48KHz through SRC (Sample Rate Conversion).
If the word length of the PCM signal is lower than that of the DAC, the Controller will automatically shift the PCM signal so that its MSB (Most Significant Bit) is aligned, and the low bit is filled with 0. If the word length of the PCM signal is higher than that of the DAC , Then you must first reduce the word length through Dither (jitter) or directly transmit it to the Codec through the AC-Link interface. If the DAC word length is not high enough for the AC-Link interface, then it will automatically increase the AC-Link interface beyond the word length The LSBs (Least Significant Bit) are removed. The DAC output is a ladder-like or pulse-like signal, and it must be filtered and reshaped by LPF (Low Pass Filter, low-pass filter) to restore the original audio signal.