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* The Speex audio compression library uses a procedure named Improved Minima Controlled Recursive Averaging, which uses a smoothed representation of spectral power and then looks at the minima of a smoothed periodogram.
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Speex and audio
The audio layer is most commonly provided by the music-oriented Vorbis format but other options include the human speech compression codec Speex, the lossless audio compression codec FLAC, and OggPCM.
Speex is a patent-free audio compression format designed for speech and also a free software speech codec that may be used on VoIP applications and podcasts.
The Speex designers see their project as complementary to the Vorbis general-purpose audio compression project.
There is already a large base of applications supporting the Speex codec, from streaming applications like teleconference ( e. g. TeamSpeak ; many servers prefer Speex due to its good quality ), to VoIP systems ( e. g. Asterisk ), to videogames ( e. g. Xbox Live, Civilization 4 ) and audio processing applications.
The media type for Speex is audio / ogg while contained by Ogg, and audio / speex ( previously audio / x-speex ) when transported through RTP or without container.
Speex can be decoded or converted to any format unlike Nellymoser audio, which was the only speech format in previous versions of Flash Player.
* CELT An audio codec patterned loosely after Speex.
* Ogg ( a container for Vorbis, FLAC, Speex ( audio formats ) & Theora ( a video format ), by the Xiph. Org Foundation )
These include Speex, an audio codec designed for speech, and FLAC, a lossless audio codec.
It was initially designed to provide subtitles for Ogg Theora videos, but is also useful for song lyrics with Ogg FLAC or Vorbis, transcripts with Ogg Speex, or any other place where it is useful to combine text with audio or video.
* Norwegian version, ePub and audio books in Speex format available.
Speex and compression
Speex is targeted at Voice over IP ( VoIP ) and file-based compression.
Speex is used as the voice compression algorithm in the Siri voice assistance on the iPhone 4S.
Xiph. org is behind other free compression formats such as Vorbis, Theora and Speex.
Speex and uses
The United States Army's Land Warrior system, designed by General Dynamics, also uses Speex for VoIP on an EPLRS radio designed by Raytheon.
The Rockbox project uses Speex for its voice interface.
The Vernier LabQuest handheld data acquisition device for science education uses Speex for voice annotations created by students and teachers using either the built-in or an external microphone.
The JavaSonics ListenUp voice recorder uses Speex to compress voice messages that are recorded in a browser and then uploaded to a web server.
This article uses material from the Speex Codec Manual which is copyright © Jean-Marc Valin and licensed under the terms of the GFDL.
Speex and .
The Speex project is an attempt to create a free software speech coder, unencumbered by patent restrictions.
** G. 722, G. 722. 1, Speex, IP-MR and others for VoIP and videoconferencing
Speex claims to be free of any patent restrictions and is licensed under the revised ( 3-clause ) BSD license.
Speex is a lossy format, meaning quality is permanently degraded to reduce file size.
The Speex project was created on February 13, 2002.
The first development versions of Speex were released under LGPL license, but as of version 1. 0 beta 1, Speex is released under Xiph's version of the ( revised ) BSD license.
Speex 1. 0 was announced on March 24, 2003, after a year of development.
The last stable version of Speex encoder and decoder is 1. 1. 12.
Xiph. Org considers Speex obsolete ; its successor is Opus.
Since Speex was designed for Voice over IP ( VoIP ) instead of cell phone use, the codec must be robust to lost packets, but not to corrupted ones.
All this led to the choice of Code Excited Linear Prediction ( CELP ) as the encoding technique to use for Speex.
; Sampling rate: Speex is mainly designed for three different sampling rates: 8 kHz ( the same sampling rate to transmit telephone calls ), 16 kHz, and 32 kHz.
audio and compression
* mpc-Musepack or MPC ( formerly known as MPEGplus, MPEG + or MP +) is an open source lossy audio codec, specifically optimized for transparent compression of stereo audio at bitrates of 160 – 180 kbit / s.
Level compression is not to be confused with audio data compression, where the amount of data is reduced without affecting the amplitude of the sound it represents.
In most amplifiers a reduction in gain takes place before hard clipping occurs ; the result is a compression effect, which ( if the amplifier is an audio amplifier ) sounds much less unpleasant to the ear.
In contrast, audio codecs for recording or broadcast can use high-latency audio compression techniques to achieve higher fidelity at a lower bit-rate.
* A device applying audio data compression to an audio signal
* A device applying dynamic range compression to an audio signal
In lossy audio compression, methods of psychoacoustics are used to remove non-audible ( or less audible ) components of the signal.
Compression of human speech is often performed with even more specialized techniques, so that " speech compression " or " voice coding " is sometimes distinguished as a separate discipline from " audio compression ".
Different audio and speech compression standards are listed under audio codecs.
Voice compression is used in Internet telephony for example, while audio compression is used for CD ripping and is decoded by audio players.
Audio data compression, as distinguished from dynamic range compression, has the potential to reduce the transmission bandwidth and storage requirements of audio data.
Audio compression algorithms are implemented in software as audio codecs.
Lossy audio compression algorithms provide higher compression at the cost of fidelity, are used in numerous audio applications.
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