在定点DSP上实现高精度独立于说话人的连续语音识别器

Y. Gong, Yu-Hung Kao
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引用次数: 22

摘要

连续语音识别是一种资源密集型算法。商用听写软件需要超过10mb的磁盘安装和32mb的RAM来运行应用程序。一个典型的嵌入式系统无法负担这么多的内存,因为它的高成本和功耗;它也没有磁盘来存储大量的静态数据(例如声学模型)。我们一直致力于优化一个适合在16位定点DSP上实现的小词汇语音识别器。该识别器支持复杂的连续密度,捆绑混合高斯,并行模型组合和噪声鲁棒的话语检测算法。定点版本与浮点版本的性能相同。该算法在100 MHz, 16位定点德州仪器TMS320C5410上实时运行,即使在驾驶条件下使用免提麦克风进行最具挑战性的连续数字拨号。
本文章由计算机程序翻译,如有差异,请以英文原文为准。
Implementing a high accuracy speaker-independent continuous speech recognizer on a fixed-point DSP
Continuous speech recognition is a resource-intensive algorithm. Commercial dictation software requires more than 10 Mbytes to install on the disk and 32 Mbytes RAM to run the application. A typical embedded system can not afford this much RAM because of its high cost and power consumption; it also lacks disk to store the large amount of static data (e.g. acoustic models). We have been working on optimization of a small vocabulary speech recognizer suitable for implementation on a 16-bit fixed-point DSP. This recognizer supports sophisticated continuous density, tied-mixtures Gaussians, parallel model combination, and a noise-robust utterance detection algorithm. The fixed-point version achieves the same performance as the floating-point version. The algorithm runs real-time on a 100 MHz, 16-bit, fixed-point Texas Instruments TMS320C5410 even for the most challenging continuous digit dialing with hands-free microphone in driving conditions.
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