Audio is a common function in electronic products. There are also a variety of products related to audio playback on the market, such as smart speakers, smart TVs and so on. This article will show you the key technologies related to audio product design, and relevant interactive audio solution.
Audio codecs are the key technology of audio products
In the design of audio products, audio codecs are a very important key technology. They are a device or software that can be used to encode or decode audio digital data streams. They can execute algorithms to compress and decompress digital audio data based on specific audio files or streaming media audio coding format. In particular, the design goal of the algorithm is to use the fewest bits in the digital audio to represent the high-fidelity audio while ensuring a certain sound quality, which can effectively reduce the bandwidth and storage space occupied by the audio file transmission process.
At present, there are quite a variety of audio codecs on the market, such as aptX, ATRAC, AAC, MP3, SBC, Opus, and other different standards and technologies. Depending on different applications, some audio codecs are designed for high-quality audio playback, most of which adopt lossless compression technology and produce a high bit rate and have high requirements for processor performance. Some are designed for network audio streaming, which seek low latency, low bit rate, and have low requirements for processor performance. In addition, some audio codec technologies are exclusively owned by certain manufacturers. If you want to adopt them, you need to be licensed by these manufacturers and pay royalties. Some audio codecs belong to open source technologies and can be used freely without royalties. Therefore, the cost of adoption is low.
Low latency audio codec for real-time interactive communication
This article will first show you the Opus audio codec, a lossy audio coding format, which aims to effectively encode voice and general audio in a single format, while maintaining low-latency enough for real-time interactive communication, and low-complexity enough for low-end embedded processors.
Opus is an open, royalty free, and versatile audio codec that can be used in different types of applications, such as voice and music streaming or compressed audio storage. It is an RFC 6716 standard formulated by the Internet Engineering Task Force (IETF). It can range from 6 kbit/s low bit rate narrowband voice to 510 kbit/s high-quality stereo music. It has low complexity scalability, making it suitable for a wide range of interactive applications.
The structure of Opus is mainly composed of two layers, one based on linear prediction (LP) and the other based on modified discrete cosine transform (MDCT). Opus effectively combines lossless and lossy technical results. For example, in speech applications, LP technologies such as code-excited linear prediction (CELP) can encode low frequencies more effectively than transform domain technologies such as MDCT. The Opus codec is composed of SILK and CELT coding technology. The former (originally developed by Skype) uses linear predictive coding (LPC), while the latter (from Xiph.Org) is completely modeled based on MDCT transform. This versatility allows Opus to run in three modes (SILK, CELT, or hybrid mode) and supports multiple configurations for different applications.
Opus can handle a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even remote live music performances. It can scale from low bitrate narrowband speech to very high quality stereo music. Supported features are: bitrates from 6 kb/s to 510 kb/s, sampling rates from 8 kHz (narrowband) to 48 kHz (fullband), frame sizes from 2.5 ms to 60 ms, constant bit rate (CBR) and variable bit rate (VBR), audio bandwidth from narrowband to fullband, and voice and music applications. It supports mono and stereo and up to 255 channels (multi stream frames), has the ability of dynamically adjustable bit rate, audio bandwidth, and frame size, has good packet loss robustness and packet loss concealment (PLC) technology, and can support floating-point and fixed-point implementation.
Highly integrated low power audio codec solutions
Arrow Electronics has introduced the Opus interactive audio codec reference solution for interactive audio applications such as smart speakers. Using STMicro STM32WB, external PA, and FP-AUD-BVLINKWB1, Arrow Electronics builds a reference design and uses advanced Opus compression algorithm technology to perform remote full-duplex voice streaming for stereo music streaming over BLE.
The STM32WB is a multiprotocol wireless and ultra-low-power device embed, a powerful and ultra-low-power radio compliant with the Bluetooth®; Low Energy SIG specification 5.2 and with IEEE 802.15.4-2011. They contain a dedicated Arm®Cortex®-M0+ for performing all the real-time low layer operation. The devices are designed to be extremely low-power and are based on the high-performance Arm® Cortex®-M4 32-bit RISC core operating at a frequency of up to 64 MHz. This core features a Floating point unit (FPU) that supports all Arm®single-precision data-processing instructions and data types. It also implements a full set of DSP instructions and a memory protection unit (MPU) that enhances application security.
Provides an overall solution with reference hardware and software design
Arrow Electronics’ team has integrated STM32WB external PA solutions into the FP-AUD-BVLINKWB1 STM32Cube function pack to perform remote full-duplex voice streaming for stereo music streaming over BLE. The system can transmit 2.4GHz signal and output 20dBm, and provides an overall solution with reference hardware and software design.
FP-AUD-BVLINKWB1 includes drivers and middleware for BLE and digital MEMS microphones. It also includes the complete Opus audio codec (v 1.3) as third-party middleware to perform bidirectional and simultaneous audio streaming between two STM32WB. The peripheral module can also communicate in full-duplex mode (bidirectional audio at 16 kHz) with a mobile device running the STBLESensor app, or receive stereo music at 48 kHz from the same app. The software with the suggested combination of STM32WB and ST devices can be used to develop wireless audio communication systems for smart home or wearable applications. The Opus algorithm used in this solution provides the flexibility to achieve high audio quality even at low bitrates, and the STM32WB has the low power capabilities to allow the development of applications featuring very low consumption.
Conclusion
The rapid growth of markets such as smart speakers and smart TVs has driven the demand for interactive audio processing. The interactive audio codec reference design solution introduced in this article will be the best choice for developing relevant products. For more relevant information, please contact the local sales personnel of Arrow Electronics.