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| Developer(s) | The Conversations Network |
|---|---|
| Stable release | 2.1.1 |
| Operating system | Windows, Mac, Linux |
| Type | Audio Editing Software |
| License | Freeware |
| Website | Levelator Homepage |
The Levelator is a software application that makes adjustments to audio signals.
The Levelator was a free application distributed by The Conversations Network [1] and developed by Bruce and Malcolm Sharpe, Norman Lorrain and Doug Kaye. Originally distributed by GigaVox Media, Inc (a for-profit company), the rights were transferred to The Conversations Network (a California 501(c)(3)) in 2008. The underlying code was originally used only for The Conversations Network's own podcasts but was subsequently released to the public, free for commercial and non-commercial use. It was unveiled to the public at the first Podcast and New Media Expo in 2005. The adjustments and drag-and-drop workflow of the Levelator make it a valuable tool for professional and non-professional broadcasters and podcasters.
As of the end of 2012, the Levelator is no longer supported or being updated by The Conversations Network. Conversations Network ceased daily operations at the end of 2012. [2]
When OS X 10.11 (El Capitan) was released, the Levelator was found to be incompatible. The original development team, Bruce Sharpe, Norman Lorrain and Doug Kaye collaborated in November 2015 to develop an OS X-only compatible release 2.1.2. [3]
In June 2020, a 64-bit, macOS Catalina version of The Levelator was revived and released for free by The Conversations Network in the Mac App Store. [4] [ better source needed ]
The Levelator adjusts the audio levels within an audio segment by combining traditional discrete compression, normalization and limiting processing. [5] By taking a global view of the data in various time segments (both long and short), the Levelator automatically balances various audio levels, such as multiple microphone levels in an interview or panel discussion, or segments combined from multiple sessions that were recorded at different levels. The Levelator can read and process PCM audio files of many sample rates and resolutions.
The Levelator reads the original audio file and creates a new audio file with balanced levels and a uniform overall volume level that is then saved in the same format as the original, but with ".output " added to the file name. Only PCM audio source files are supported (most major file formats, including WAV and AIFF [6] ). Video and lossy compressed audio are not supported, encouraging use of The Levelator at the correct point in the production chain - i.e. before lossy encoding to the delivery format such as MP3. [7]
An audio file format is a file format for storing digital audio data on a computer system. The bit layout of the audio data is called the audio coding format and can be uncompressed, or compressed to reduce the file size, often using lossy compression. The data can be a raw bitstream in an audio coding format, but it is usually embedded in a container format or an audio data format with defined storage layer.
The Au file format is a simple audio file format introduced by Sun Microsystems. The format was common on NeXT systems and on early Web pages. Originally it was headerless, being simply 8-bit μ-law-encoded data at an 8000 Hz sample rate. Hardware from other vendors often used sample rates as high as 8192 Hz, often integer multiples of video clock signal frequencies. Newer files have a header that consists of six unsigned 32-bit words, an optional information chunk which is always of non-zero size, and then the data.
A codec is a device or computer program that encodes or decodes a data stream or signal. Codec is a portmanteau of coder/decoder.
In information theory, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compression reduces bits by identifying and eliminating statistical redundancy. No information is lost in lossless compression. Lossy compression reduces bits by removing unnecessary or less important information. Typically, a device that performs data compression is referred to as an encoder, and one that performs the reversal of the process (decompression) as a decoder.
Ogg is a free, open container format maintained by the Xiph.Org Foundation. The authors of the Ogg format state that it is unrestricted by software patents and is designed to provide for efficient streaming and manipulation of high-quality digital multimedia. Its name is derived from "ogging", jargon from the computer game Netrek.
Speex is an audio compression codec specifically tuned for the reproduction of human speech and also a free software speech codec that may be used on VoIP applications and podcasts. It is based on the CELP speech coding algorithm. Speex claims to be free of any patent restrictions and is licensed under the revised (3-clause) BSD license. It may be used with the Ogg container format or directly transmitted over UDP/RTP. It may also be used with the FLV container format.
Audio Interchange File Format (AIFF) is an audio file format standard used for storing sound data for personal computers and other electronic audio devices. The format was developed by Apple Inc. in 1988 based on Electronic Arts' Interchange File Format and is most commonly used on Apple Macintosh computer systems.
FLAC is an audio coding format for lossless compression of digital audio, developed by the Xiph.Org Foundation, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to between 50 and 70 percent of its original size and decompresses to an identical copy of the original audio data.
Monkey's Audio is an algorithm and file format for lossless audio data compression. Lossless data compression does not discard data during the process of encoding, unlike lossy compression methods such as Advanced Audio Coding, MP3, Vorbis, and Opus. Therefore, it may be decompressed to a file that is identical to the source material.
DVD-Audio is a digital format for delivering high-fidelity audio content on a DVD. DVD-Audio uses most of the storage on the disc for high-quality audio and is not intended to be a video delivery format.
Xiph.Org Foundation is a nonprofit organization that produces free multimedia formats and software tools. It focuses on the Ogg family of formats, the most successful of which has been Vorbis, an open and freely licensed audio format and codec designed to compete with the patented WMA, MP3 and AAC. As of 2013, development work was focused on Daala, an open and patent-free video format and codec designed to compete with VP9 and the patented High Efficiency Video Coding.
Sound quality is typically an assessment of the accuracy, fidelity, or intelligibility of audio output from an electronic device. Quality can be measured objectively, such as when tools are used to gauge the accuracy with which the device reproduces an original sound; or it can be measured subjectively, such as when human listeners respond to the sound or gauge its perceived similarity to another sound.
The Apple Lossless Audio Codec (ALAC), also known as Apple Lossless, or Apple Lossless Encoder (ALE), is an audio coding format, and its reference audio codec implementation, developed by Apple Inc. for lossless data compression of digital music. After initially keeping it proprietary from its inception in 2004, in late 2011 Apple made the codec available open source and royalty-free. Traditionally, Apple has referred to the codec as Apple Lossless, though more recently it has begun to use the abbreviated term ALAC when referring to the codec.
Transcoding is the direct digital-to-digital conversion of one encoding to another, such as for video data files, audio files, or character encoding. This is usually done in cases where a target device does not support the format or has limited storage capacity that mandates a reduced file size, or to convert incompatible or obsolete data to a better-supported or modern format.
WavPack is a free and open-source lossless audio compression format and application implementing the format. It is unique in the way that it supports hybrid audio compression alongside normal compression which is similar to how FLAC works. It also supports compressing a wide variety of lossless formats, including various variants of PCM and also DSD as used in SACDs, together with its support for surround audio.
Rockbox is a free and open-source software replacement for the OEM firmware in various forms of digital audio players (DAPs) with an original kernel. It offers an alternative to the player's operating system, in many cases without removing the original firmware, which provides a plug-in architecture for adding various enhancements and functions. Enhancements include personal digital assistant (PDA) functions, applications, utilities, and games. Rockbox can also retrofit video playback functions on players first released in mid-2000. Rockbox includes a voice-driven user-interface suitable for operation by visually impaired users.
A portable media player (PMP) is a portable consumer electronics device capable of storing and playing digital media such as audio, images, and video files. The data is typically stored on a compact disc (CD), Digital Video Disc (DVD), Blu-ray Disc (BD), flash memory, microdrive, or hard drive; most earlier PMPs used physical media, but modern players mostly use flash memory. In contrast, analogue portable audio players play music from non-digital media that use analogue media, such as cassette tapes or vinyl records.
OpenShot Video Editor is a free and open-source video editor for Windows, macOS, Linux, and ChromeOS. The project started in August 2008 by Jonathan Thomas, with the objective of providing a stable, free, and friendly to use video editor.
An audio coding format is a content representation format for storage or transmission of digital audio. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.
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