Multi-Rate Audio Codec Filtering for Low-Latency Signal Mixing
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Solution Overview
Problem
Conventional audio processing systems face challenges in efficiently handling multi-rate audio signals from various sources, particularly in wireless devices, where they need to support duplex operations and adapt to new communication and multimedia technologies, requiring advanced codec devices with flexible processing capabilities.
Innovation Solution
A method and system for processing multi-rate audio signals that involves up-sampling audio signals using half-band interpolation, dynamically adjusting gain, multi-band equalization, and selecting finite impulse response (FIR) filters to ensure compatibility and quality, with the option to substitute upsample-by-two filters with IIR filters to reduce latency.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Manufacturing precision
If audio signals from multiple sources with different sampling rates are processed using conventional methods, then the system can maintain simplicity, but the processing quality and adaptability to multi-rate signals deteriorate
Solution Approach 1:
The audio processing system is divided into separate functional modules: a sampling rate conversion module that handles rate adaptation, an audio processing module that performs filtering and mixing, and a output module that generates final audio signals. This segmentation allows each module to be optimized independently for quality while managing overall system complexity.
Solution Approach 2:
The audio codec device is designed with universal processing capabilities that can handle multiple audio sources with different sampling rates (e.g., 8kHz, 16kHz, 44.1kHz, 48kHz) through a common sampling rate conversion architecture. The system can adapt to various audio formats and sources without requiring separate dedicated processing paths for each rate.
2Adaptability or versatility
If the system supports duplex operations with multiple audio sources, then the adaptability and functionality improve, but the processing complexity and computational requirements increase
Solution Approach 1:
The audio processing system dynamically adjusts processing parameters based on the characteristics of input signals. The sampling rate conversion and filtering operations are adapted in real-time according to the detected audio source types and their respective sampling rates, allowing the system to optimize performance for each specific configuration rather than using fixed complex processing for all cases.
Solution Approach 2:
A sampling rate conversion module acts as an intermediary between multiple audio sources with different rates and the audio processing module. This intermediary component standardizes the sampling rates of incoming signals before they enter the main processing pipeline, simplifying subsequent processing operations and reducing overall system complexity.
3Manufacturing precision
If upsampling filters are used to convert different sampling rates to a common rate, then the audio quality is improved, but the processing latency increases
Solution Approach 1:
Different filtering approaches are applied to different audio sources based on their specific requirements. For example, voice signals may use simpler filtering with lower latency, while music signals may use more sophisticated filtering for higher quality. This localized optimization allows the system to achieve good audio quality where needed without incurring unnecessary latency for all signals.
Solution Approach 2:
The system can dynamically change filtering parameters such as filter order and cutoff frequencies based on the processing requirements. When low latency is prioritized, the system uses lower-order filters with fewer computational steps. When maximum quality is required, higher-order filters are employed. This parameter adjustment allows flexible trade-off between quality and latency.
Data Source
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AI summary
An audio codec in a wireless device may be utilized for up sampling two or more audio signals to a same data sampling rate. Each audio signal, such as digital audio, voice, and polyringer, for example, may be received at one of a plurality of data sampling rates. Audio signals may be equalized and/or compensated with an FIR filter before up sampling or with an IIR filter to reduce overall processing latency. Multiple half-band interpolation operations may perform the up sampling. The first half-band filter may be replaced by aw IIR filter to reduce overall processing latency. A gain of the up-sampled data may be adjusted to reduce noise effects. The channels of the up-sampled audio signals may be mixed and later further up sampled for subsequent communication to an output device. The up-sampled mixed audio signals may be down sampled for communication to a Bluetooth radio.