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[Other resourcead_pcm

Description: 用在语音芯片上的一个,实现ADPCM编码的压缩程序,对语音编解码有帮助!-with the voice of a chip to achieve ADPCM compression coding procedures, the voice codecs help!
Platform: | Size: 3114 | Author: 张三 | Hits:

[Communication2400 bps MELP语音编解码器浮点算法

Description: 2400 bps MELP语音编解码器浮点算法,一种高压缩比的另类语音压缩算法,可以供学习和开发借鉴使用-MELP 2400 bps voice codecs floating-point algorithm, a high compression ratio of alternative voice compression algorithm, for learning and development from the use of
Platform: | Size: 762432 | Author: 安迪 | Hits:

[Communication2400 bps MELP语音编解码器定点算法

Description: 2400 bps MELP语音编解码器定点算法,一种先进的另类语音压缩算法,可以供学习和借鉴使用-MELP 2400 bps voice codecs fixed-point algorithms, an advanced alternative voice compression algorithm, for it to learn from the use of
Platform: | Size: 146048 | Author: 安迪 | Hits:

[VOIP programspeex-1.0.5.tar

Description: 一个非常好的开源音频编解码项目,支持多种音频采用频率,支持多码流,支持可变速率 Speex a free codec for free speech Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications by providing a free alternative to expensive proprietary speech codecs. Moreover, Speex is well-adapted to Internet applications and provides useful features that are not present in most other codecs. Finally, Speex is part of the GNU Project and is available under the Xiph.org variant of the BSD license-a very good open-source audio codecs, support for multiple audio frequency, multi-stream, Variable-rate support for a free Vorbis codec for free speech Vorbis i s an Open Source / Free Software patent-free Aud io compression format designed for speech. The Vorbis Project aims to lower the barrier of entry for voice applications by providing a free alte rnative to expensive proprietary speech codec s. Moreover, Vorbis is well-adapted to the Internet application s and provides useful features that are not pres ent in most other codecs. Finally, Vorbis is part of the GNU Project and is available under the Xiph.org variant of the BSD license
Platform: | Size: 546872 | Author: 邹远富 | Hits:

[Speech/Voice recognition/combine16to8K-Downsample

Description: 本程序将指定的16K采样的语音数据文件转换为经G.723编解码后的8K语音数据。降采样前先使用180阶的FIR滤波器对语音数据进行频率压缩,然后进行抽取,并对抽取的数据进行G.723编解码。该程序在非特定语音识别的库文件处理中使用,也可扩展至其他用途。-this procedure will be designated the 16K sampling voice data files converted to G.723 codecs by the 8K words Music data. Sampling down before use 180 bands FIR filter frequency voice data compression, then proceed to collect, also collected data G.723 codecs. The procedures in non-specific voice recognition for the use of document processing, and can be expanded to other uses.
Platform: | Size: 1675652 | Author: 王小飞 | Hits:

[Other resourceDSP公司内部使用的

Description: 公司内重要价值的dsp压缩代码,强烈推荐!!!含(G711压缩解压,VAD算法语音激活算法)-company value of the DSP code compression and strongly recommended! ! ! Containing (G711 codecs, voice-activated VAD algorithm algorithm)
Platform: | Size: 15538 | Author: 后来居 | Hits:

[DSP programDSP公司内部使用的

Description: 公司内重要价值的dsp压缩代码,强烈推荐!!!含(G711压缩解压,VAD算法语音激活算法)-company value of the DSP code compression and strongly recommended! ! ! Containing (G711 codecs, voice-activated VAD algorithm algorithm)
Platform: | Size: 15360 | Author: 后来居 | Hits:

[Algorithmad_pcm

Description: 用在语音芯片上的一个,实现ADPCM编码的压缩程序,对语音编解码有帮助!-with the voice of a chip to achieve ADPCM compression coding procedures, the voice codecs help!
Platform: | Size: 3072 | Author: 张三 | Hits:

[Voice CompressG.723.1-Annex-A-199611-I

Description: ITU-T G.723语音压缩编解码算法源码和测试语音,内有标准附件和使用说明。-ITU-T G.723 voice compression codec algorithm source code and test voice, within the standard accessories and use.
Platform: | Size: 1700864 | Author: 阙劲峰 | Hits:

[Communication2400 bps MELP语音编解码器浮点算法

Description: 2400 bps MELP语音编解码器浮点算法,一种高压缩比的另类语音压缩算法,可以供学习和开发借鉴使用-MELP 2400 bps voice codecs floating-point algorithm, a high compression ratio of alternative voice compression algorithm, for learning and development from the use of
Platform: | Size: 761856 | Author: 安迪 | Hits:

[Communication2400 bps MELP语音编解码器定点算法

Description: 2400 bps MELP语音编解码器定点算法,一种先进的另类语音压缩算法,可以供学习和借鉴使用-MELP 2400 bps voice codecs fixed-point algorithms, an advanced alternative voice compression algorithm, for it to learn from the use of
Platform: | Size: 146432 | Author: 安迪 | Hits:

[VOIP programspeex-1.0.5.tar

Description: 一个非常好的开源音频编解码项目,支持多种音频采用频率,支持多码流,支持可变速率 Speex a free codec for free speech Speex is an Open Source/Free Software patent-free audio compression format designed for speech. The Speex Project aims to lower the barrier of entry for voice applications by providing a free alternative to expensive proprietary speech codecs. Moreover, Speex is well-adapted to Internet applications and provides useful features that are not present in most other codecs. Finally, Speex is part of the GNU Project and is available under the Xiph.org variant of the BSD license-a very good open-source audio codecs, support for multiple audio frequency, multi-stream, Variable-rate support for a free Vorbis codec for free speech Vorbis i s an Open Source/Free Software patent-free Aud io compression format designed for speech. The Vorbis Project aims to lower the barrier of entry for voice applications by providing a free alte rnative to expensive proprietary speech codec s. Moreover, Vorbis is well-adapted to the Internet application s and provides useful features that are not pres ent in most other codecs. Finally, Vorbis is part of the GNU Project and is available under the Xiph.org variant of the BSD license
Platform: | Size: 546816 | Author: 邹远富 | Hits:

[Speech/Voice recognition/combine16to8K-Downsample

Description: 本程序将指定的16K采样的语音数据文件转换为经G.723编解码后的8K语音数据。降采样前先使用180阶的FIR滤波器对语音数据进行频率压缩,然后进行抽取,并对抽取的数据进行G.723编解码。该程序在非特定语音识别的库文件处理中使用,也可扩展至其他用途。-this procedure will be designated the 16K sampling voice data files converted to G.723 codecs by the 8K words Music data. Sampling down before use 180 bands FIR filter frequency voice data compression, then proceed to collect, also collected data G.723 codecs. The procedures in non-specific voice recognition for the use of document processing, and can be expanded to other uses.
Platform: | Size: 1675264 | Author: 王小飞 | Hits:

[Windows DevelopG.711G.721G.723encodeanddecode

Description: 音频编解码G.711G.721声音压缩编解码-Audio CODECs G.711G.721 voice compression codec
Platform: | Size: 95232 | Author: libaokun | Hits:

[Linux-UnixHawkVoiceDI091src

Description: HawkVoice is a game oriented, multiplayer voice over network API released under the GNU Library General Public License (LGPL) , with support for Linux® /UNIX® systems and Windows® 9x/ME/NT/2000/XP/CE. It is designed to be a portable, free, open source code alternative to the Microsoft® DirectPlay® Voice in DX8-9. It provides voice compression using several free voice codecs. The very low bitrate (VLB) codecs, those less than 6 Kbps, are optimized for the compression of human speech.-HawkVoice is a game oriented, multiplayer voice over network API released under the GNU Library General Public License (LGPL) , with support for Linux® /UNIX® systems and Windows® 9x/ME/NT/2000/XP/CE. It is designed to be a portable, free, open source code alternative to the Microsoft® DirectPlay® Voice in DX8-9. It provides voice compression using several free voice codecs. The very low bitrate (VLB) codecs, those less than 6 Kbps, are optimized for the compression of human speech.
Platform: | Size: 477184 | Author: afc | Hits:

[SCM05-music-coding

Description: Music Coding  LPC-based codecs model the sound source to achieve good compression.  Works well for voice.  Terrible for music.  What if you can’t model the source?  Model the limitations of the human ear.  Not all sounds in the sampled audio can actually be heard.  Analyze the audio and send only the sounds that can be heard.  Quantize more coarsely where noise will be less audible.-Music Coding  LPC-based codecs model the sound source to achieve good compression.  Works well for voice.  Terrible for music.  What if you can’t model the source?  Model the limitations of the human ear.  Not all sounds in the sampled audio can actually be heard.  Analyze the audio and send only the sounds that can be heard.  Quantize more coarsely where noise will be less audible.
Platform: | Size: 1226752 | Author: vidhyarthi | Hits:

[Voice Compressgsm-1.0-pl13

Description: gsm语音编解码,多种压缩速率支持,C语言实现-gsm voice codecs, support for multiple compression rate, C language
Platform: | Size: 87040 | Author: xie | Hits:

[Compress-Decompress algrithmsspeex-1.2

Description: 是一套主要针对语音的开源免费,无专利保护的音频压缩格式。Speex工程着力于通过提供一个可以替代高性能语音编解码来降低语音应用输入门槛 。另外,相对于其它编解码器,Speex也很适合网络应用,在网络应用上有着自己独特的优势。同时,Speex还是GNU工程的一部分,在改版的BSD协议中得到了很好的支持(It is an open source, free, patent free audio compression format for speech. Speex works to reduce the threshold for voice applications by providing an alternative to high-performance speech codecs. In addition, compared with other codecs, Speex is also suitable for network applications, and has its own unique advantages in network applications. At the same time, Speex is part of the GNU project and has been well supported in the revised BSD protocol)
Platform: | Size: 1949696 | Author: 啊所到之处 | Hits:

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