Multi-Rate Audio Up-Sampling With Low-Latency Filter Switching
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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 systems like mobile phones, where they need to support duplex operations and new communication technologies, requiring advanced codec devices with flexible and efficient processing capabilities.
Innovation Solution
A system and method for processing multi-rate audio signals involve up-sampling audio signals to a common data sampling rate using FIR or IIR filters, adjusting gain to reduce noise, and employing half-band interpolation, with dynamic filtering and equalization to support multiple wireless standards like WCDMA, GSM, and Bluetooth, enabling efficient mixing and communication to output devices.
Engineering Contradictions & Design Principles
Engineering Contradiction Analysis
1Measurement precision
If audio signals are up-sampled using traditional filtering methods, then audio quality is improved, but processing latency increases
Solution Approach 1:
The patent implements dynamic filtering where the filter characteristics are adjusted based on the input signal properties and processing requirements. This allows the system to optimize between audio quality and latency by adapting filter parameters in real-time rather than using fixed filtering approaches
Solution Approach 2:
The up-sampling process is divided into multiple stages with different filtering approaches. Half-band filters are used in specific stages to reduce computational complexity while maintaining audio quality, and processing is segmented to allow early output of less critical samples while prioritizing critical path processing
2Adaptability or versatility
If multiple audio processing sources are supported, then system versatility is improved, but device complexity increases
Solution Approach 1:
The patent implements a universal audio processing architecture that can handle multiple audio sources and formats through a common up-sampling and mixing framework. The system supports various wireless standards (WCDMA, GSM, Bluetooth) and audio formats through configurable processing paths rather than dedicated hardware for each source type
Solution Approach 2:
The system manages complexity by dynamically changing processing parameters such as sampling rates, filter coefficients, and mixing ratios based on the active audio sources and their requirements. This allows flexible support for multiple sources without requiring complex fixed infrastructure for each possible configuration
3Measurement precision
If gain adjustment is applied to reduce noise, then signal quality is improved, but processing time increases
Solution Approach 1:
Gain adjustment and noise reduction are performed as part of the up-sampling process itself rather than as separate post-processing steps. The filtering and gain adjustment operations are integrated into the interpolation stages, allowing noise reduction to occur concurrently with rate conversion without adding sequential processing delays
Data Source
AI summary
One or more circuits in a mobile phone 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 and one or more of the following wireless standards: WCDMA, HSDPA, GSM, GPRS, EDGE, and/or Bluetooth. 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 an 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.