Audio interface in portable applications

Different digital audio subsystems have been used to create interfaces for digital conversion between several microprocessors or DSPs (digital signal processors) and audio devices. This article describes several audio interface specifications that are currently available in the market.

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PCM specification

One of the simplest audio interfaces is the so-called PCM (Pulse Code Modulation) interface. Strictly speaking, all digital signals are transmitted through the PCM and require careful reference to the mono mechanism for digital phones. The PCM interface consists of a clock pulse (BCLK), a frame sync signal (FS), and a data queue, each PCM corresponding to a data to be received or to be transmitted.

On the rising edge of the FS signal, the data transmission begins with the MSB (Most SignificantBit) word and the FS frequency is equal to the sampling rate. The transmission of the data word begins after the FS signal, a single data bit is transmitted in sequence, and one data word is transmitted in one clock cycle. When the MSB is sent, the level of the signal is first minimized to avoid loss of the MSB when different interface schemes are used on different terminals. In addition to the RJ (Right-justified) format that is fading in applications, this approach is currently used in most audio interfaces.

The PCM interface is easy to implement and can in principle support any data scheme and any sample rate, but requires a separate data queue for each audio channel. This property makes PCM extremely popular in primary target applications such as digital phones. s Choice.

I2S specification

The I2S interface (Inter-IC Sound) was first used by Philips in consumer audio in the 1980s and multiplexed in a signal mechanism called LRCLK (Left/RightCLOCK) to combine two audio signals into a single data. queue. When LRCLK is high, the left channel data is transmitted; when LRCLK is low, the right channel data is transmitted. Compared to PCM, I2S is more suitable for stereo systems. For multi-channel systems, it is also possible to execute several data queues in parallel under the same BCLK and LRCLK conditions.

However, Hi-Fi audio requirements in portable systems are higher than stereo. First, more complex audio ICs are usually controlled by writing to internal registers. Since I2S, PCM, and similar audio interfaces do not provide register entries, separate control interfaces are required, such as increasing the number of pins of the audio IC on the controller. Second, the ability to perform audio at different sample rates is critical, with 44.1 kHz (standard frequency for audio CDs) and 48 kHz (computer audio standard frequencies) for a very wide range of frequencies.

For I2S and its derivatives, the system either generates Low-jitterBCLK and LRCLK at different frequencies (or FS in PCM) or converts all audio streams to a single sample rate in a software environment. . The first case requires at least one analog phase-locked loop (PLL) and two synchronous feedbacks to be recorded on different frequencies. And when evaluating the power of an interface, the increased power consumption must be factored into. The second case, while enhancing computing power, also significantly increases the power consumption of the processor. And when this processor executes the user application at the same time, the entire system runs slower and even stops when the audio is turned on.

More and more consumers now want better audio and more features from digital cameras, digital camcorders, MP3 players, portable DVD players, portable multimedia players and multimedia processors. Wolfson has expanded the scope of the decoder for this purpose. The recently launched WM897x system I2S audio protocol not only improves the system integration, but also improves the system's audio quality.

Take the stereo WM8980 and the mono WM8982 as an example, you can connect directly to the TV through the on-screen display. In portable systems, the same video features as high-quality audio require an additional video amplifier, and the WM8978 system is an upgrade to the full stereo capability of the WM8974 system introduced in October 2004.

The three products use the DSP microprocessor as the core, which can filter the wind and so on to improve the recording function of the audio system, especially in the visualization system. In addition, the new product also uses a 5-band and 3D audio system to improve the audio output and programmable resistance filter to eliminate noise. These systems also typically support microphones and cell phone speakers with clock frequencies between 12MHz and 19MHz, further reducing the number of components in the product. To meet the high-quality audio speakers and piezoelectric speakers, the power consumption can reach 900mW. The digital recording playback limiter prevents excessive output of the speakers. The analog part of the three decoder products requires a supply voltage as low as 2.5V. The supply voltage is as low as 1.6V.

With the dramatic increase in the demand for mobile products, by combining the two interface technologies, it is possible to combine the form of simple transmission of mono audio with an expandable standard such as Hi-Fi function, which greatly improves the combination. Battery life and integration of resources, such as the ability to easily handle incoming calls during MP3 playback.

WM8753L integrates the stereo Hi-Fi analog-to-digital converter in IIS protocol with independent mono PCM digital-to-analog conversion technology in one chip, and has the digital-to-analog conversion function of both IIS protocol and PCM interface, which makes MP3 , conversations and other audio features can work together.

The MAX8753L is one of the few decoders that incorporates PCM/Hi-Fi-enabled analog sections with operating voltages below 1.8V and digital sections operating below 1.42V. At 1.8V operating voltage, the decoder consumes 7mW for stereo playback and the minimum power consumption for PCM is less than 6mW. The system integrates dual interface technology for connecting different loudspeakers, including the speakers, headphones and the drive section of the handset. The external device no longer requires a separate earphone or headphone amplifier section, and the cap-less interface mode can connect all loads. The embedded digital signal processing system can control tone, bass boost, auto-adjust headphone volume or analog-to-digital converter. These two analog-to-digital conversion methods enable noise cancellation or stereo storage of dual DSP systems.

The WM8753L Hi-Fi analog-to-digital converter can be used as a control part or as a system with a USB interface with a main clock frequency of 12MHz to 24MHz, a 19.2MHz mobile system, and a standard 256fs ratio of 12.288MHz and 24.576MHz. Controlled part. Its internal phase-reduced loop system produces the clock frequencies needed to meet PCM and Hi-Fi conversions. If the required clock frequency in the audio system already exists, the phase-locked loop can be used for other purposes. AC'97/AC-Link Specifications

The AC'97 (Audio Coding 1997) standard was specified by Intel Corporation for computer audio. Unlike PCM and I2S, AC'97 is not just a data format, it is used for internal coding specifications of audio coding, it also has control functions. The well-known AC-Link interface includes a bit clock (BITCLK), sync signal correction (SYNC), and a data queue from the code to the processor and from the slave (SDATDIN and SDATAOUT). The AC'97 data frame starts with a SYNC pulse and consists of 12 20-bit time segments (time segments are different destination services defined in the standard) and 16-bit "tag" segments for a total of 256 data sequences. For example, time periods "1" and "2" are used to access the encoded control registers, while time periods "3" and "4" load the left and right audio channels, respectively. The "tag" segment indicates which of the other segments contains valid data. Dividing the frame into time segments makes it possible to transmit control signals and to reach only 9 audio channels or to other data streams through only 4 lines. Compared to the I2S solution with a separate control interface, AC'97 significantly reduces the overall pin count.

For example, playing audio in the 44.1 kHz band, each time period is marked as invalid at one frame of more than 12 frames, and the effective data points are evenly distributed through the D/A converter in the encoder to form low distortion for each time period. Analog signal, this method has considerably less power consumption than PLL or with sample rate conversion.

The complexity of AC'97 lies in the higher number of gates and the power consumption of the interface itself. Through system-level measures such as built-in multi-rate power supply, AC'97 still consumes a lot of power, so AC'97 is suitable for using more than one sample. Rate of complex systems such as telephones and multimedia PDAs. Its inherent 20-bit data solution and sample rate up to 48 kHz are very rare in portable applications where battery life and small size are as important as audio quality.

Unlike I2S, AC'97 has a unique bandwidth and transport protocol when transmitting audio-free additional data codes. Therefore, there is no need to add an additional digital touch screen when the AC'97 system is in use. Wolfson has used this feature to provide integrated touch screen interface technology, on-chip display driver technology, high-fidelity stereo technology, sound and ring tone management technology for systems such as the WM9712. Pen write detection and pressure detection capabilities can be used to transfer data between the audio system and the PDA portable system using a 4-pin AC-link bus and data interface.

This integration of AC'97 and PCM systems into micro PCs, PDAs and smartphones is similar to the I2S/PCS system in terms of maximizing battery life. Wolfson's WM9713 product adds an audio decoder to the WM9712 for mobile phone conversation management to maximize battery life.

Azalia specifications into computer and consumer audio

In computer and consumer audio, the AC'97 specification is being replaced by the Azalia specification recently developed by Intel. This new standard is an enhancement to the AC'97 specification, which includes a 32-bit solution with a sampling rate of up to 192 kHz, flexible configuration of input/output pins and headphones attached to the socket sensor, activation into a single socket Speaker. In addition, the extra power consumption caused by the bit clock of 27.576 kHz (which is twice that of AC'97) is a major obstacle to extending battery life. Therefore, if you do not consider development in other markets, Azalia has little chance of becoming mainstream in portable applications.

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