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Description: ogg 相关代码, 游戏声音编码一种非常棒的格式， 一种高质量的有损压缩格式, 被广泛应用在游戏声音引擎里-ogg-related code, the game a great voice encoding format, a high-quality lossy compression format, has been widely used in game sound engine
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Size: 478208 |
Author: 熊 |
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Description: java在网页中操作摄像头的源码,质量清晰,可进行语音-java in the page source to operate the camera, the quality of clarity, it can conduct voice
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Size: 10240 |
Author: gl474c1 |
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Description: 我搜集资料研究开发的人声过滤算法，可以把立体声音乐中的歌词和音乐分离开来。效果一般，过滤后，有的音乐质量略有下降。供大家参考。供测试的数据文件，没包含在包中，太大了。-I have collected information on the research and development of the human voice filtering algorithm, can be stereo music lyrics and music separate. General effect, filter, some slight decline in music quality. For your reference. For the test data files, not included in the package, too.
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Size: 156672 |
Author: hyeewang |
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Description: 音频编码，能够将声音格式进行压缩，保持比较好的质量-Audio coding, voice formats can be compressed to maintain a relatively good quality
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Size: 1328128 |
Author: hell |
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Description: 这个网络电话程序是linux下，用C语言实现的。它既不是实现的H.323 或 SIP协议, 也没有使用RTP协议，更没有使用到任何其它第三方软件，不过，它确实工作的很好。通话话音质量非常不错。-The network telephone program is under linux, using C language realization. It is not the realization of the H.323 or SIP protocol, but also did not use the RTP protocol, but did not use to any other third-party software, but it does work well. Voice call quality is very good.
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Size: 53248 |
Author: 严锐 |
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Description: ADPCM(Adaptive Differential Pulse Code Modulation)，是一种针对 16bits( 或8bits或者更高) 声音波形数据的一种有损压缩算法,它将声音流中每次采样的 16bit 数据以 4bit 存储,所以压缩比 1:4. 而且压缩/解压缩算法非常简单,所以是一种低空间消耗，高质量高效率声音获得的好途径。保存声音的数据文件后缀名为 .AUD 的大多用ADPCM 压缩。-ADPCM (Adaptive Differential Pulse Code Modulation), is a response to 16bits (or 8 bits or higher) voice waveform data of a lossy compression algorithm, it will sound each stream sampling 16bit data to 4bit storage, so the compression ratio 1:4. and compression/decompression algorithm is very simple, there is a low space consumption, high-quality sound and efficient way to get good. Preservation of the voice data file is named suffix. AUD mostly using ADPCM compression.
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Size: 2048 |
Author: 王磊 |
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Description: C51语音播放源码
将语音按占空比放出。原语音为8位8KHz，则125us一个字节，现时钟主频近2MHz，周期为0.5us，这样一个字节占250个周期，而字节8位为256，可以近似为256个周期，实验应放在定时器中产生。
如果倍频，每个字节就可以产生两个波形，音质应更好
-C51 voice broadcast voice-source release by the duty cycle. The original voice for the 8-bit 8KHz, while 125us a byte, it is near the clock frequency 2MHz, cycle 0.5us, accounting for a 250-byte cycle, and 8 bytes for the 256, can be approximately 256 cycles, the experiment produced should be placed on timers. If the octave, each byte can have two waveforms, sound quality should be better
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Size: 20480 |
Author: 啊买噶的 |
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Description: 用于语音通信质量测量的国际标准代码PESQ-Measuring the quality of voice communications for international standards code PESQ
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Size: 38912 |
Author: 刘辉 |
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Description:
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Size: 6144 |
Author: 武月 |
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Description: 1.MPEG-2 audio 系統的software，可解出MPEG-2的聲音
2.支援MPEG-2的高質量聲音但不包含AAC decoder-1.MPEG-2 audio system software, the solvability of the MPEG-2 s voice 2. Support MPEG-2 high-quality sound but does not include the AAC decoder
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Size: 625664 |
Author: 孫小明 |
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Description: 语音芯片的驱动程序，1700是替换isd1420 isd25系列的最新产品，音质性能都有提高。isd1730 isd1760全部兼容。如果做语音芯片的话-Voice chip driver, 1700 is the replacement series isd1420 isd25 the latest products, sound quality has to improve performance. isd1730 isd1760 all compatible. If so, then voice chip
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Size: 137216 |
Author: 齐磊 |
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Description: 这是一个使用 lpc2103 做的语音播放器,特别之处在于采用了pwm将16位数字信号转换为模拟信号,可以达到CD的播放音质,带全部源码.-This is a voice so use LPC2103 player, so special is the use of the pwm will be 16-bit digital signal is converted to analog signals, can achieve the CD player sound quality, with full source code.
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Size: 692224 |
Author: 越飞越高 |
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Description: ISD 语音芯片的驱动程序。 原有产品已经停产，isd1700作为替代产品，功能强大，音质略好于其isd14xx系列等。-ISD voice chip driver. The original product has been discontinued, isd1700 as an alternative product, powerful, slightly better sound quality in its isd14xx series.
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Size: 36864 |
Author: 刘于 |
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Description: 高质量源码语音通话代码VC,基于windowsP平台-High-quality voice calls source code VC, based on the platform windowsP
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Size: 664576 |
Author: zhang |
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Description: 1、 支持实时声音采集，保存格式可以为WAV或MP3
2、 支持实时电平显示。
3、 支持音量大小侦测。
4、 支持采样频率（44100，22050，11025，8000等），声道（单，双）的选择。
5、 支持声卡的选择，当有多块声卡时，可以容易选择一块声卡。
6、 支持编程语言：DELPHI，VC，VB，CB，VFP，PB等编程语言。
7、 音质,品质绝佳，行内领先技术。
-1, supports real-time voice of the collection, preservation format can be WAV or MP3 2, supports real-time-level display. 3, to support the size of the detection volume. 4, support the sampling frequency (44100,22050,11025,8000, etc.), mono (single, double) choice. 5, support the choice of sound card, when the number of pieces of sound card, you can easily select a sound card. 6, support for programming languages: DELPHI, VC, VB, CB, VFP, PB and other programming languages. 7, sound quality, excellent-quality, leading technology firms.
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Size: 754688 |
Author: mdsong |
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Description: 根据TMS320C55XX系列的DSP在语音信号数字处理方面的强大能力，本文介绍一种采用这种 DSP与高性能立体声音频解码芯片TLV320 AIC23相结合的MP3解码器系统的设计。实验证明该系统可以顺利实现MP3数据流的上传与下载，高质量完成 MP3的解码与播放 。-According to TMS320C55XX series of DSP in the digital voice signal processing capabilities, this paper introduces a DSP with the use of such high-performance stereo audio decoder chip TLV320 AIC23 combining MP3 decoder system design. Experiments show that the system can successfully upload MP3 data streams and downloads, high quality MP3 decoder and playback.
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Size: 161792 |
Author: 凝雪 |
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Description: 本文提出了一种基于SIP协议的软件电话的软件结构和设计实现方案。该方案以嵌入式Windows CE为平台，中间件采用开源的SIP协议栈oSIP/eXosip，通过协议栈的移植和在协议栈之上应用程序的开发，实现了SIP软件电话。测试结果表明，该软件电话在布置Windows CE的PDA上具有良好的语音通话质量。 -In this paper, a software-based SIP-phone software architecture and design to achieve the program. The program is embedded Windows CE as a platform, using open-source middleware SIP protocol stack oSIP/eXosip, through the protocol stack of transplantation and in the protocol stack on top of application development and achieving a SIP software phone. Test results show that the layout of the software phone in the PDA on Windows CE has a good voice call quality.
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Size: 146432 |
Author: 杰哥 |
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Description: Mumble是一款针对游戏玩家的低延迟，高质量的音频通信工具，可以让玩家在游戏的同时进行实时的语音交流。软件支持消除回音，这样在游戏过程中，其它玩家就不会听到用户喇叭里发出的游戏声音了。-Mumble is a low latency for gamers, high-quality audio communication tools, allows gamers to play at the same time communicate in real-time voice. Software support eliminate echo, so that the course of the game, other players would not hear the user issued loudspeakers sound the Games.
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Size: 1169408 |
Author: 花花 |
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Description: 目前市场上有很多语音录放系统，如录放音玩具、录音笔等，大多采用了单片机控制一个语音芯片，再接一个FLASH存储器的结构。单片机可以控制录放时间，选取特定时间段的播放以及单多声道的录放，容易通过改变外接存储FLASH改变录放时间。
我们做的是一个简单模型，只能录放。
只需要录音和放音时的外部 ROSC 端振荡电阻不同就能改变声音的录入和播放速度，录入的时间越短音质越好，录入的时间越长音质越差
它采用模拟量直接存贮技术，因此保真度高，音质好
-Currently on the market there is a lot of voice recording systems, such as sound recording toys, sound recording pens and so on, most of the single-chip microcomputer to control the use of a voice chip, making the structure of a FLASH memory. Single-chip microcomputer can control the playback time, select a specific period of time many of the players, as well as single-channel recording, easy to FLASH memory by changing the external playback time change. We make a simple model, only playback. Required only when the recording and playback-side oscillation of the external resistor ROSC will be able to change different recording and playback of sound speed, the shorter the time of admission, the better sound quality, longer recording sound worse using direct analog storage technology, Thus the high-fidelity, better sound quality
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Size: 306176 |
Author: liusixue |
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Description: DSB-SC信号的生成与解调
1) 用离散（DSP）的方法生成DSB信号
2) 载波频率为150KHz，音频为500Hz和2000Hz的混合音。
3) 加入高斯白噪声
（4） 语音信号
的传输。
改变抽样频率和量化台阶大小，观察重建信号以及量化噪声信号的波形；对于语音信号主观评价声音质量的变化。
-DSB-SC signal generation and demodulation 1) Discrete (DSP) methods to generate DSB signal 2) the carrier frequency to 150KHz, audio 500Hz and 2000Hz for the mixed tone. 3) by adding white Gaussian noise (4) transmission of voice signals. Change the sampling frequency and quantization step size, to observe the reconstruction of signal and quantization noise signal waveform for subjective evaluation of the speech signal changes in sound quality.
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Size: 28672 |
Author: Ryan |
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