exhale, which is an acronym for “Ecodis eXtended High-efficiency And Low-complexity Encoder”, is a lightweight library and application to encode uncompressed WAVE-format audio files into MPEG-4 format files complying with the ISO/IEC 23003-3 (MPEG-D) Unified Speech and Audio Coding (USAC, also known as Extended High-Efficiency AAC) standard. In addition, exhale writes program peak-level and loudness data into the generated MPEG-4 files according to the ISO/IEC 23003-4, Dynamic Range Control (DRC) specification for use by decoders providing DRC. exhale currently makes use of all frequency-domain (FD) coding tools in the scale factor based MDCT processing path, except for predictive joint stereo, which is still being integrated. Its objective is high quality mono, stereo, and multichannel coding at medium and high bit rates, so the lower-rate USAC coding tools (ACELP, TCX, Enhanced SBR and MPEG Surround with Unified Stereo coding) won’t be integrated.
A Mac OSX GUI frontend for open source audio codecs. There is an older Windows version available as well.
WavPack is a completely open audio compression format providing lossless, high-quality lossy, and a unique hybrid compression mode. For version 5.0.0, several new file formats and lossless DSD audio compression were added, making WavPack a universal audio archiving solution.
In the default lossless mode WavPack acts just like a WinZip compressor for audio files. However, unlike MP3 or WMA encoding which can affect the sound quality, not a single bit of the original information is lost, so there’s no chance of degradation. This makes lossless mode ideal for archiving audio material or any other situation where quality is paramount. The compression ratio depends on the source material, but generally is between 30% and 70%.
The hybrid mode provides all the advantages of lossless compression with an additional bonus. Instead of creating a single file, this mode creates both a relatively small, high-quality lossy file that can be used all by itself, and a “correction” file that (when combined with the lossy file) provides full lossless restoration. For some users this means never having to choose between lossless and lossy compression!
WavPack employs only well known, public domain techniques (i.e., linear prediction with LMS adaptation, Elias and Golomb codes) in its implementation. Methods and algorithms that have ever been patented (e.g., arithmetic coding, LZW compression) are specifically avoided. This ensures that WavPack encoders and decoders will remain open and royalty-free.
JAMin is the JACK Audio Connection Kit (JACK) Audio Mastering interface. JAMin is an open source application designed to perform professional audio mastering of stereo input streams. It uses LADSPA for digital signal processing (DSP). JAMin is licensed under the GPL.
- Linear filters
- JACK I/O
- 30 band graphic EQ
- 1023 band hand drawn EQ with parametric controls
- Spectrum analyser
- 3 band peak compressor
- Lookahead brickwall limiter
- Multiband stereo processing
- Presets and scenes
- Loudness maximizer
Levelator® adjusts the audio levels within your podcast or other audio file for variations from one speaker to the next, for example. It’s not a compressor, normalizer or limiter although it contains all three. It’s much more than those tools, and it’s much simpler to use. The UI is dirt-simple: Drag-and-drop any WAV or AIFF file onto The Leveler’s application window, and a few moments later you’ll find a new version which just sounds better.
Vorbis is a free and open-source software project headed by the Xiph.Org Foundation. The project produces an audio coding format and software reference encoder/decoder (codec) for lossy audio compression. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis. ~ en.wikipedia.org/wiki/Vorbis
Ogg Vorbis is a completely open, patent-free, professional audio encoding and streaming technology with all the benefits of Open Source.
VorbisGain is a utility that uses a psychoacoustic method to correct the volume of an Ogg Vorbis file to a predefined standardized loudness.
LAME is used to encode / compress audio data into the lossy MP3 file format. It’s a high quality MPEG Audio Layer III (MP3) encoder, licensed under the LGPL.
LAME Downloads ~ Rarewares
LAME Recommended Settings – HA Wiki
LAME Libraries – RareWares
LAME Source Code ~ Sourceforge
lame3995o ~ A fork of LAME (alternative version)
LameVST ~ LAME compression as a VST effect
Wikipedia ~ LAME info
WinLAME ~ Windows front-end (GUI)
wxlame ~ Windows front-end (GUI)