Comprehending I2S Protocol

In this article, we’ll delve into the I2S protocol, its technical specifications, and explore more about how it helps in achieving ultra-low latency.

Audio systems communication and their requirements have changed drastically as a result of rapid technological evolution. As a result, they have become more digitalized. In most systems, these digital audio signals are typically processed by devices such as DSPs, DACs, or ADCs. Despite being innovative, digitalization also invites certain challenges like audio delay which influences the quality of experience.

Audio delay is like a hiccup – a brief pause between generating and hearing a sound when transmitted from one component to another. Although imperceptible, it often disrupts the natural flow of the audio experience. Latency is the measure of this delay, and the lower the latency, the better the experience.

One effective method for overcoming these challenges and achieving low latency is the Inter-IC Sound (I2S) protocol – a specialized communication protocol for audio data. In this article, let us explore the I2S protocol, where we break down its technical specification and explain what makes I2S ideal for achieving ultra-low latency in digital audio systems.

The I2S Protocol: An Overview  

The Inter-IC Sound (I2S) protocol is a synchronous serial communication protocol designed explicitly for transmitting audio data. This synchronous nature ensures ultra-low latency by maintaining precise timing between sender and receiver. At its core, I2S also provides high-quality audio transfer without compression, making it an ideal choice for applications demanding pristine sound reproduction. This forms the basis of digital audio communication in various digital devices across telecommunications, finance, multimedia streaming, cloud computing, healthcare, and autonomous systems.

The I2S protocol communication consists of three vital signals:

• Bit Clock (BCLK):

This signal provides the timing reference for the data transfer, synchronizing communication between the transmitting and receiving devices. It runs continuously and controls the rate at which data bits are transmitted and received.

• Frame Sync (FSYNC):

FSYNC marks the start of a new audio sample within the data stream, allowing receivers to identify each audio frame’s beginning.  It also indicates which audio channel (left or right) the data pertains to.

• Serial Data (SDOUT):

This signal carries the actual audio data in serial format. Optionally, an additional serial data channel (named SDOUT1) may also be present.

How Does I2S Work?

I2S delivers audio data continuously without interruptions or fixed-size frames, showcasing its seamless flow and flexible data alignment. This versatility makes I2S a protocol capable of handling various audio formats, instilling confidence in its adaptability.

1. Data Synchronization: 

The data on the SDOUT line is synchronized with the BCLK, ensuring precise timing for data read/write operations. Data bits are sampled on the clock signal’s rising or falling edge depending on the configuration.

2. Word Framing: 

The FSYNC signal marks the beginning of a new audio word and changes state to indicate whether the data pertains to the left or right channel. This transition aligns the data correctly with the audio channel.

3. Continuous Data Stream: 

I2S transmits data continuously without needing fixed-size frames. The MSB of one audio word is directly followed by the MSB of the next. This uninterrupted flow of data is crucial for high-quality audio transmission.

4. Data Alignment:

The protocol allows flexible data alignment, supports varying data lengths, and accommodates different audio formats.

5.  Word Length and Format:

While I2S does not restrict word length, the transmitter and receiver must agree on the format to ensure correct data interpretation. Any extra bits are ignored, or if fewer bits are received, they are padded with zeros.

I2S Configurations

To accommodate various system architectures, I2S offers different configuration options. These configurations determine the roles and responsibilities of the devices involved in the audio data transfer process. The I2S has three configurations in general,

• Transmitter as Master: The transmitter generates the clock (BCLK) and FSYNC signals, which the receiver uses to synchronize data reading.
• Receiver as Master: The receiver generates the clock (BCLK) and FSYNC signals. The transmitter uses these signals to synchronize and send the data on the SOUT line.
• Controller as Master: This independent controller generates the clock and FSYNC signals. The transmitter and receiver use these signals to synchronize their data transmission and reception.

What Makes I2S Ideal for Achieving Ultra-Low Latency in Digital Audio Systems? 

I2S is meticulously engineered to minimize latency at every stage of the audio signal path. By optimizing data transfer, clocking, and processing, it also delivers exceptional performance for applications demanding real-time audio. Let’s delve into the attributes that contribute to I2S’s low-latency prowess.

• Serial Data Transmission  

I2S’s serial data transmission reduces wiring complexity and minimizes data skew, which can occur in parallel systems. This method, synchronized by a dedicated bit clock line, ensures precise timing and low latency, providing a practical and efficient solution for audio systems.

• Dedicated Clock Lines  

I2S uses a bit clock and a frame sync clock to control the data transmission. The bit clock determines the transition from one bit to another, while the frame sync clock delineates the audio samples. This synchronization allows for accurate timing and reduces the potential delays that can occur with asynchronous systems.

• Minimal Processing Overhead  

The I2S protocol is designed to minimize processing overhead and reduce latency in audio data transmission. Key aspects include:

→ High Sample Rates: I2S supports high sample rates, which can further reduce the delay between audio samples. This capability allows for more precise timing and lower latency, although this requires careful system resource management.

→ Direct Memory Access (DMA): Using DMA for audio data transfer helps to lower CPU overhead and latency. DMA allows data to be transferred directly between peripherals and memory, reducing the need for CPU intervention and improving efficiency.

→ Minimal Buffering and Processing: I2S is optimized to work with minimal buffering and processing overhead, directly streaming audio data between devices without intermediate buffering.

• Embedded Systems Optimization  

I2S is well-suited for embedded systems with limited computational resources. The protocol’s efficiency in handling audio data transmission with minimal processing ensures minimal latency, even in constrained environments.

 Real-World Customary Applications of I2S  

I2S is widely used in various applications that require low latency, high-fidelity audio transmission. Some of these applications include:

• Consumer Electronics: Devices like smartphones, tablets, and digital televisions use I2S for high-quality audio streaming.
• Professional Audio Equipment: Mixing consoles, audio interfaces, and other professional audio gear rely on I2S for real-time audio processing.
• Automotive Systems: Modern car infotainment systems use I2S to provide high-quality, low-latency audio to passengers.

Enhance Your TDM and I2S Protocol Compliance Validation with Soliton  

Digital audio devices often support various protocols to offer developers greater flexibility. Some audio devices feature TDM interfaces for audio data communication.

Enter Soliton’s expertise in audio protocol validation. Our comprehensive validation services for I2S and TDM protocols make achieving ultra-low latency seamless. And, with our validation expertise, make your audio devices deliver the superior audio experience demanded by today’s high-performance applications.

Written by

Dinesh Babu and Adam Jayahar B

August 7, 2024