主题：[求助]期盼各位专家的指点.万分感谢!!!

MP3是音频数字信号的一种格式,因是一种(有损伤)压缩编码,占据的存储空间小,受到大众的欢迎.但是在放音时,仍然要经过数模(D/A)变换这个环节,信号成为模拟的了,再经音频放大,再通过耳机或扬声器放出.音乐或语音信号本身是模拟信号,目前为方便使用,要转换为数字信号,MP3比WAVE是有损伤的,但是信号的频率成分基本上没变化.就是说,原来是16KHZ的SIN声波,录音后,变成MP3格式,由合适的放音设备(比如MP3播放器)放出的自然仍然是16KHZ的SIN声波.也许问题会出在录音时的传声器和放音时的扬声器的性能是不是能保证16KHZ的频率响应了.

我在1990年的冬天,在JOHNS HOPKINS大学的电机系访问过一个数字信号处理的专家和他的一个研究生.

他们的一个课题是对新生儿的听觉作早期诊断.

就是通过耳机给新生儿发出一个脉冲声波,然后用传感器检取新生儿的脑电波並记录下来,来判断新生儿的听觉.

我给他们的建议是:

1.把接受到的脑电波的信号作FFT分析,看新生儿的听觉的频镨特征.

2.给新生儿发出一个脉冲声波的强度怎么定?总不能伤害新生儿啊.

他们很欣赏我的第一个建议,对第二个问题,说他们是很小心的,不会伤及孩子,而且这个项目是和JOHNS HOPKINS医院的有关专家的合作课题.

希望上述的看法对你有帮助.

啊,你可能的问题是要次声发生器啊.

中科院声学研究所和南京大学早就有这方面的研究啊.

4HZ的纯音,哪家的扬声器行呢?

我以为理论上是做得了的,但是,为了做出几个实验样品,计划经济时代要看你这个项目是谁下达的.

现在是市场经济,哪个厂会做呢?我想业余爱好者会有兴趣做的.

MP3能不能重放16KHZ的信号?

请看:

http://www.iis.fraunhofer.de/amm/techinf/layer3/index.html

引用上面网页的有关内容.

History

In 1987, the Fraunhofer IIS started to work on perceptual audio coding in the framework of the EUREKA project EU147, Digital Audio Broadcasting (DAB). In a joint cooperation with the University of Erlangen (Prof. Dieter Seitzer), the Fraunhofer IIS finally devised a very powerful algorithm that is standardized as ISO-MPEG Audio Layer-3 (IS 11172-3 and IS 13818-3).

Without data reduction, digital audio signals typically consist of 16 bit samples recorded at a sampling rate more than twice the actual audio bandwidth (e.g. 44.1 kHz for Compact Discs). So you end up with more than 1.400 Mbit to represent just one second of stereo music in CD quality. By using MPEG audio coding, you may shrink down the original sound data from a CD by a factor of 12, without losing sound quality. Factors of 24 and even more still maintain a sound quality that is significantly better than what you get by just reducing the sampling rate and the resolution of your samples. Basically, this is realized by perceptual coding techniques addressing the perception of sound waves by the human ear.

Using MPEG audio, one may achieve a typical data reduction of

1:4	by Layer 1 (corresponds to 384 kbps for a stereo signal),
1:6...1:8	by Layer 2 (corresponds to 256..192 kbps for a stereo signal),
1:10...1:12	by Layer 3 (corresponds to 128..112 kbps for a stereo signal),

still maintaining the original CD sound quality.

By exploiting stereo effects and by limiting the audio bandwidth, the coding schemes may achieve an acceptable sound quality at even lower bitrates. MPEG Layer-3 is the most powerful member of the MPEG audio coding family. For a given sound quality level, it requires the lowest bitrate - or for a given bitrate, it achieves the highest sound quality.

Sound Quality

Some typical performance data of MPEG Layer-3 are:

sound quality	bandwidth	mode	bitrate	reduction ratio
telephone sound	2.5 kHz	mono	8 kbps *	96:1
better than short wave	4.5 kHz	mono	16 kbps	48:1
better than AM radio	7.5 kHz	mono	32 kbps	24:1
similar to FM radio	11 kHz	stereo	56...64 kbps	26...24:1
near-CD	15 kHz	stereo	96 kbps	16:1
CD	>15 kHz	stereo	112..128kbps	14..12:1
*) Fraunhofer IIS uses a non-ISO extension of MPEG Layer-3 for enhanced performance ("MPEG 2.5")

In all international listening tests, MPEG Layer-3 impressively proved its superior performance, maintaining the original sound quality at a data reduction of 1:12 (around 64 kbit/s per audio channel). If applications may tolerate a limited bandwidth of around 10 kHz, a reasonable sound quality for stereo signals can be achieved even at a reduction of 1:24.

For the use of low bit-rate audio coding schemes in broadcast applications at bitrates of 60 kbit/s per audio channel, the ITU-R recommends MPEG Layer-3. (ITU-R doc. BS.1115)

看完以上介绍,可以说,就原定的指标,即使是按CD质量,16KHZ的重放没有保证.

好,问题明确了,是16KHZ的重放.

用WAV格式会好的多.另外可以用SINE信号发生器,在网上可找到这个软件,

然后声卡的输出口那儿就引出常规的音频模拟信号,经放大器(20HZ-20KHZ频宽的),馈给扬声器.

关于扬声器,能发16KHZ已不是难题,更高一些也没问题,这儿有大批的扬声器专家,请他们给你具体建议吧.

信号源可访问

http://www.trueaudio.com

他们的True RTA 带信号源的分析仪很好.

当然花钱可买到全功能的.

FREE的也可以用,信号源的频率选择是跳跃的.