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The ASRC-Lite is a 16-bit audio sample rate converter designed to manage multiple audio channels simultaneously. It achieves a total harmonic distortion plus noise (THD+N) of -90dB, maintaining audio integrity across different devices with varying sample rates. The core supports asynchronous sample rate conversion, ensuring seamless audio synchronization between diverse equipment without compromising sound quality.
The ASRC-Pro elevates multi-channel audio processing with 24-bit precision, achieving an impressive THD+N of -130dB. It's tailored for professional audio environments that demand superior audio fidelity and conversion accuracy. This core facilitates the integration of audio systems operating at various sample rates, ensuring high-quality audio output across different platforms.
The ASRC-Lite-up provides advanced audio conversion capabilities, particularly for systems requiring audio sample rate upgrades. Maintaining a THD+N of -90dB, it delivers high-quality audio conversion for multi-channel setups. This converter is vital for applications where audio equipment must be synchronized despite varied input and output rates.
Compatible with MPEG-1/2 Layer I/II audio standards, this decoder is suitable for devices requiring compatibility with older audio formats. It processes multiple streams efficiently, ensuring playback quality and format versatility in various multimedia devices.
This decoder supports AAC-LC and HE-AAC audio formats, providing real-time decoding capabilities. It's designed to handle multiple audio streams, ensuring high-quality playback for applications such as media players and broadcast systems.
The ASRC-Premier serves as a formidable multi-channel audio sample rate converter with a 20-bit depth and a THD+N of -120dB. It is targeted at high-end audio applications that necessitate precise audio conversion and synchronization. This IP is optimal for achieving superior audio standards in environments where maintaining sound fidelity is crucial across various media formats and devices.
This encoder offers real-time audio compression, supporting both AAC-LC and HE-AAC formats. It's designed for applications needing efficient streaming and storage of high-quality audio. It can be deployed on FPGAs or ASICs and supports multi-stream processing, making it versatile for various audio broadcasting environments.
Focused on decoding AAC-LC format audio, this core supports real-time processing for seamless audio experiences. Ideal for integration into digital audio products, it ensures quality sound reproduction across numerous consumer electronics devices.
Tailored for encoding audio in the AAC-LC format, this core ensures high compression efficiency and audio fidelity. It serves applications ranging from streaming services to digital broadcasting, offering flexibility in managing varying audio quality requirements across different platforms.
Optimized for stereo audio encoding in the AAC-LC format, this encoder supports efficient bitrate management while preserving sound quality. Its customizable architecture is particularly beneficial for applications where high-quality audio is essential, such as digital music distribution and high-definition broadcasts.
Designed to encode audio into MPEG-1/2 Layer I/II formats, this encoder is ideal for legacy systems and environments requiring specific audio standards. It supports real-time processing for multiple streams, making it a robust solution for various multimedia applications that require format compatibility and efficient bandwidth use.
This decoder provides support for both MPEG-1/2 and AAC formats, delivering versatile audio playback options. It's suitable for multiple-stream processing, ensuring audio fidelity across a wide range of consumer and professional applications.
This interface effectively manages asynchronous sample rate conversions between different audio domains. Supporting I2S and TDM formats, it ensures high-fidelity audio synchronization without requiring related input/output clock signals, proving vital for diverse digital audio equipment.
SPDIF-Rx-Pro serves as a configurable receiver for SPDIF-AES3 formats, supporting multiple audio channels. It's a fully digital design allowing seamless integration into existing audio systems without needing external PLLs, making it ideal for broadcast and recording environments.
The I2S/TDM-Rx+Tx-Pro is a transceiver that manages audio signals over I2S and TDM channels. It offers flexibility in handling bi-directional audio data, making it imperative for systems that demand simultaneous audio receiving and transmitting capabilities.
The I2S/TDM-Rx-Pro is a configurable multi-channel serial audio receiver supporting standard audio interfaces like I2S and TDM. Its programmable nature allows for easy adaptation to different audio processing and professional recording setups, enhancing compatibility across systems.
This transmitter supports multi-channel audio transmission over I2S and TDM interfaces. It enables precise synchronization and integration into audio systems that require robust transmission standards, making it suitable for professional audio and studio environments.
This transmitter handles SPDIF-AES3 audio streams, providing reliable signal transmission across various interface types. Designed with full digital technology, it's an excellent choice for systems looking for high-quality audio data transfer without external components like PLLs.
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