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speex-1.0.5.tar
- 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
yuyingbianmajiema
- 语音编码＿解码代码库（C语言）.已调试. 基于DSP-_ Speech Coding codec code library (C language). Debugging has. DSP-based
amr_SOUCCE
- ANSI C code for the Adaptive Multi Rate (AMR) speech codec-ANSI C code for the Adaptive Multi Rate (AMR ) speech codec
ts_126073v060000p0
- GSM AMR-NB speech codec R98 Version 7.6.0 December 12, 2001 R99 Version 3.3.0 REL-4 Version 4.1.0 REL-5 version 5.3.0 REL-6 version 6.0.0 -GSM AMR-NB speech codec R98 Version 7.6.0 D ecember 12, 2001's R99 Version 3.3.0 REL-4 Version 4.1.0
G721-G723
- G721-G723 源代码 G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for
G728(LD-CELP)
- G.728 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s
G729(CS-ACELP)
- G.729 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s
vic
- vic语音编解码标准，应用于voip系统-speech codec standard applied voip system
The_Speex_Codec_Manual_Version_1.2_Beta_3
- Speex是一套开源的专门压缩声音的库，压缩的性能非常高，常用在VoIP或者其它网络程序中。Speex声称自己是不受任何专利限制，并授权根据修订后的BSD许可证发布。它可以用来与Ogg容器格式或直接在UDP / RTP协议下传输。 这份是Speex的编码手册英文版，下面的地址是维基百科中关于Speex的介绍: http://en.wikipedia.org/wiki/Speex-Speex is a free software speech codec that may be used on
qcelp8k
- 介绍北美3G通信中语音编码标准QCELP8k 文档-North American 3th G speech codec---QCELP8k spec
audiobuf
- Simple IP telephony program ussing OSS linux audio system and GSM speech codec
encoder
- Implementation of a speech codec based on coding of speech at 8 kbit/s using conjugate-structure algebraic-code-excited linear-prediction (CS-ACELP) - We took .wav files that is sampled at 8000 Hz using 16-bit linear PCM. The encoding process i
iPhone-streaming-media-player-
- 在参照了第三代合作伙伴计划的分组交换流媒体服务技术规范的基础上，介绍移动流媒体网络体系中使用的网络协议，论述H.264视频解码技术和AMR-NB(Adaptive Multi-Rate Speech Codec Narrow Band)及AAC（Advanced Audio Coding）音频解码技术以及承载音视频媒体数据的容器3GPP（3rd GenerationPartnership Project）文件结构，针对iPhone平台自身特点，重点研究在iPhone平台上移动流媒体播放器的实
b9cdc07ae6b7
- implementation of speech codec
amr-SPEECH
- AMR语音编解码程序，官网上下的是编解码分开的。这里做了修改合到一起。-AMR SPEECH CODEC VC PROJECT
26073-b00
- 3GPP TS 26.073 V11.0.0 ANSI-C AMR编码 最新版本源代码以及说明文档。 -3GPP TS 26.073 V11.0.0 ANSI-C code for the Adaptive Multi Rate (AMR) speech codec
g723
- g7231speech编解码，非常经典。源代码未优化过，适合编解码人员来学习-speech codec
g.726-speech-codec-source-program-
- g.726语音压缩编解码源程序 亲测可用-G. 726 speech codec source program pro is available
26204-800
- AMR WB+ 基于3GPP标准的宽带Adaptive Multi-Rate - Wideband 实现-3rd Generation Partnership Project Technical Specification Group Services and System Aspects Speech codec speech processing functions Adaptive Multi-Rate- Wideband (AMR-WB) speech codec
SILK
- SILK语音编码研究必备资料，包括基本原理介绍以及C语音代码总汇-This document describes SILK, a speech codec for real-time, packetbased voice communications