wanglihe / 3gpp-amrwb
amrnb codec from 3gpp official website http://www.3gpp.org/DynaReport/26204.htm
☆7Updated 10 years ago
Related projects: ⓘ
- Linphone.org mirror for opencore-amr (git://git.code.sf.net/p/opencore-amr/code)☆21Updated 3 months ago
- Source codes of SILK codec(v1.0.9)☆33Updated 6 years ago
- Resampler Module Port From WebRTC☆18Updated 6 years ago
- VisualOn AMR-WB encoder from Android☆35Updated 9 years ago
- 使用Opencore-AMR对 AMR <--> PCM 转码封装☆34Updated 9 years ago
- android opensles hostapi for portaudio☆47Updated 4 months ago
- Real-time library for sample rate conversion☆142Updated 3 weeks ago
- An ANSI C library for encoding/decoding using the A-law and u-Law.☆62Updated 12 years ago
- Acoustic Echo Cancellation builtin WebRTC aec/aecm(mobile) module, speex 1.0/1.2.☆75Updated 6 years ago
- codec for audio in G72X, G711,G723 G726 G729 and encode or decode them from PCM☆38Updated 8 years ago
- SDK文件同步☆21Updated 4 years ago
- A Simple and Efficient Audio Resampler Implementation in C☆137Updated 4 years ago
- Freeware Advanced Audio Coder faac mirror☆176Updated last year
- The ITU G.722 codec, Copyright (C) 2005 Steve Underwood, Copyright (c) CMU 1993 Computer Science, Speech Group Chengxiang Lu and Alex Hau…☆34Updated last month
- This is an audio resamplerate program based on Secret Rabbit Code and iniparser.☆22Updated 3 years ago
- webRTC aec模块 ,单独编译,相关头文件都已经整理☆39Updated 4 years ago
- Audio Loudness Normalization Filter Port From FFmpeg☆10Updated 5 years ago
- Applying webrtc's acoustic echo cancellation (AEC) to audio files☆34Updated 8 years ago
- Automatic Gain Control Module Port From WebRTC☆166Updated 5 years ago
- Voice Activity Detector Module Port From WebRTC☆158Updated 4 years ago
- Java library for speech enhancement☆28Updated 9 years ago
- WebRTC AudioProc (AEC, VAD, NS...)☆98Updated 3 years ago
- PCM to G711 Fast Conversions☆56Updated 6 years ago
- Packaged version of iLBC codec from the WebRTC project☆62Updated 7 months ago
- Speex voice codec mirror - THIS IS A MIRROR, DEVELOPMENT HAPPENS AT https://gitlab.xiph.org/xiph/speex☆430Updated 2 months ago
- Conversion of raw AMR RTP payload packets to .amr storage format☆17Updated last year
- a simple wave resampling by fft example.☆21Updated 6 years ago
- https://developer.skype.com/silk☆45Updated 8 years ago
- Low Complexity Communication Codec Plus (LC3plus)☆35Updated 3 years ago
- A Simple and Efficient Implementation Of Fast Fourier Transform For Audio Resampler☆52Updated 5 years ago