CELT 0 votes

The CELT codec is a compression algorithm for audio. Like MP3, Vorbis, and AAC it is suitable for transmitting music with high quality. Unlike these formats CELT imposes very little delay on the signal, even less than is typical for speech centric formats like Speex, GSM, or G.729.

Using CELT application developers can build software that allows musicians to perform together across the Internet, or simply build great sounding telephony systems. Why shouldn't your telephone sound as good as your stereo?

Like other open and unencumbered technology from the Xiph.Org Foundation CELT requires no royalties and no complicated licensing.

CELT stands for “Constrained Energy Lapped Transform”. It applies some of the CELP principles, but does everything in the frequency domain, which removes some of the limitations of CELP. CELT is suitable for both speech and music and currently features:

1. Ultra-low latency (typically from 5 to 22.5 ms)
2. Full audio bandwidth (≥20kHz; sample rates from 8 kHz to 48 kHz)
3. Support for both speech and music
4. A quality/bitrate trade-off competitive with widely used high delay codecs
5. Stereo support
6. Packet loss concealment
7. Constant bit-rates from 32 kbps to 128 kbps and above
8. A fixed-point version of the encoder and decoder

The CELT codec is meant to bridge the gap between Vorbis and Speex for applications where both high quality audio and low delay are desired.

internet Speech Audio Codec(ISAC) 0 votes

internet Speech Audio Codec (iSAC) is a wideband speech codec, developed by Global IP Solutions (GIPS) (acquired by Google Inc in 2011). It is suitable for VoIP applications and streaming audio. The encoded blocks have to be encapsulated in a suitable protocol for transport, eg. RTP.

It is one of the codecs used by AIM Triton, the Gizmo5, QQ, and Google Talk. It was formerly a proprietary codec licensed by Global IP Solutions. As of June 2011, it is part of open source WebRTC project, which includes a royalty-free license for iSAC when using the WebRTC codebase.

OpenSebJ 0 votes

OpenSebJ is a free real time audio sample mixer that runs on Microsoft Windows. The source code for OpenSebJ is released under the GNU General Public License.

Features includes:

  1. Supports loading of 255 Audio samples (wave format), which can be played in real time
  2. Sample properties such as, Volume, Pan and Frequency can be adjusted individually for each sample (even during mid play)
  3. A composition tool is provided, presented in a familiar multi-track sequencer environment. Looping compositions are also supported.
  4. Samples can be linked to keys on the keyboard and trigged in real time when a key press occurs
  5. Each sample can also have the play cursor position adjusted while the sample is playing, to provide a virtual needle to be moved
  6. Stream to disk recording, allows all audio played from OpenSebJ to be saved
  7. Pitch shifter roll, allows an order to be setup, so that each time a sample is played the next pitch shift in the sequence is undertaken before the sample is played. I.e. set up a sample to key '1', then setup a pitch shift sequence, high,mid,low,higher etc (using frequency values which are transposed to on screen 'ranges' to allow easier utilisation).