Developer(s) | LAME Developers |
---|---|
Initial release | 1998 |
Stable release | 3.100 [1] / 13 October 2017 |
Repository | sourceforge |
Operating system | Cross-platform |
Type | Codec |
License | GNU Lesser General Public License [2] |
Website | lame |
LAME is a software encoder that converts digital audio into the MP3 audio coding format. LAME is a free software project that was first released in 1998 and has incorporated many improvements since then, including an improved psychoacoustic model. The LAME encoder outperforms early encoders like L3enc [3] and possibly the "gold standard encoder" MP3enc, both marketed by Fraunhofer. [4]
LAME was required by some programs released as free software in which LAME was linked for MP3 support. This avoided including LAME itself, which used patented techniques, and so required patent licenses in some countries. All relevant patents have since expired, and LAME is now bundled with Audacity. [5]
The name LAME is a recursive acronym for LAME Ain't an MP3 Encoder. [6]
Around mid-1998, Mike Cheng created LAME 1.0 as a set of modifications against the 8Hz-MP3 encoder source code. After some quality concerns were raised by others, he decided to start again from scratch based on the dist10 MPEG reference software sources. His goal was only to speed up the dist10 sources, and leave its quality untouched. That branch (a patch against the reference sources) became Lame 2.0. The project quickly became a team project. Mike Cheng eventually left leadership and started working on tooLAME (an MP2 encoder). [7]
Mark Taylor then started pursuing increased quality in addition to better speed, and released version 3.0 featuring gpsycho, a new psychoacoustic model he developed. A few key improvements since LAME 3.x, in chronological order: [7]
Like all MP3 encoders, LAME implemented techniques covered by patents owned by the Fraunhofer Society and others. The developers of LAME did not license the technology described by these patents. Distributing compiled binaries of LAME, its libraries, or programs that derive from LAME in countries where those patents have been granted may have constituted infringement, but since 23 April 2017, all of these patents have expired. [8] [9]
The LAME developers stated that, since their code was only released in source code form, it should only be considered as an educational description of an MP3 encoder, and thus did not infringe any patent in itself. They also advised users to obtain relevant patent licenses before including a compiled version of the encoder in a product. [10] Some software was released using this strategy: companies used the LAME library, but obtained patent licenses.
In the course of the 2005 Sony BMG copy protection rootkit scandal, there were reports that the Extended Copy Protection rootkit included on some Sony compact discs had portions of the LAME library without complying with the terms of the LGPL. [11] [12] [13]
MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg. It was designed to greatly reduce the amount of data required to represent audio, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners; for example, compared to CD-quality digital audio, MP3 compression can commonly achieve a 75–95% reduction in size, depending on the bit rate. In popular usage, MP3 often refers to files of sound or music recordings stored in the MP3 file format (.mp3) on consumer electronic devices.
MPEG-1 is a standard for lossy compression of video and audio. It is designed to compress VHS-quality raw digital video and CD audio down to about 1.5 Mbit/s without excessive quality loss, making video CDs, digital cable/satellite TV and digital audio broadcasting (DAB) practical.
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, libvorbis. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis.
Adaptive Transform Acoustic Coding (ATRAC) is a family of proprietary audio compression algorithms developed by Sony. MiniDisc was the first commercial product to incorporate ATRAC, in 1992. ATRAC allowed a relatively small disc like MiniDisc to have the same running time as CD while storing audio information with minimal perceptible loss in quality. Improvements to the codec in the form of ATRAC3, ATRAC3plus, and ATRAC Advanced Lossless followed in 1999, 2002, and 2006 respectively.
Variable bitrate (VBR) is a term used in telecommunications and computing that relates to the bitrate used in sound or video encoding. As opposed to constant bitrate (CBR), VBR files vary the amount of output data per time segment. VBR allows a higher bitrate to be allocated to the more complex segments of media files while less space is allocated to less complex segments. The average of these rates can be calculated to produce an average bitrate for the file.
MPEG-1 Audio Layer II or MPEG-2 Audio Layer II is a lossy audio compression format defined by ISO/IEC 11172-3 alongside MPEG-1 Audio Layer I and MPEG-1 Audio Layer III (MP3). While MP3 is much more popular for PC and Internet applications, MP2 remains a dominant standard for audio broadcasting.
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression. It was designed to be the successor of the MP3 format and generally achieves higher sound quality than MP3 at the same bit rate.
Musepack or MPC is an open source lossy audio codec, specifically optimized for transparent compression of stereo audio at bitrates of 160–180 kbit/s. It was formerly known as MPEGplus, MPEG+ or MP+.
High-Efficiency Advanced Audio Coding (HE-AAC) is an audio coding format for lossy data compression of digital audio defined as an MPEG-4 Audio profile in ISO/IEC 14496–3. It is an extension of Low Complexity AAC (AAC-LC) optimized for low-bitrate applications such as streaming audio. The usage profile HE-AAC v1 uses spectral band replication (SBR) to enhance the modified discrete cosine transform (MDCT) compression efficiency in the frequency domain. The usage profile HE-AAC v2 couples SBR with Parametric Stereo (PS) to further enhance the compression efficiency of stereo signals.
MPEG-4 Part 2, MPEG-4 Visual is a video compression format developed by the Moving Picture Experts Group (MPEG). It belongs to the MPEG-4 ISO/IEC standards. It uses block-wise motion compensation and a discrete cosine transform (DCT), similar to previous standards such as MPEG-1 Part 2 and H.262/MPEG-2 Part 2.
FAAC is a software project which includes the AAC encoder FAAC and decoder FAAD2. It supports MPEG-2 AAC as well as MPEG-4 AAC. It supports several MPEG-4 Audio object types, file formats, multichannel and gapless encoding/decoding and MP4 metadata tags. The encoder and decoder is compatible with standard-compliant audio applications using one or more of these object types and facilities. It also supports Digital Radio Mondiale.
Extended Copy Protection (XCP) is a software package developed by the British company First 4 Internet and sold as a copy protection or digital rights management (DRM) scheme for Compact Discs. It was used on some CDs distributed by Sony BMG and sparked the 2005 Sony BMG CD copy protection scandal; in that context it is also known as the Sony rootkit.
Α video codec is software or a device that provides encoding and decoding for digital video, and which may or may not include the use of video compression and/or decompression. Most codecs are typically implementations of video coding formats.
MPEG-4 SLS, or MPEG-4 Scalable to Lossless as per ISO/IEC 14496-3:2005/Amd 3:2006 (Scalable Lossless Coding), is an extension to the MPEG-4 Part 3 (MPEG-4 Audio) standard to allow lossless audio compression scalable to lossy MPEG-4 General Audio coding methods (e.g., variations of AAC). It was developed jointly by the Institute for Infocomm Research (I2R) and Fraunhofer, which commercializes its implementation of a limited subset of the standard under the name of HD-AAC. Standardization of the HD-AAC profile for MPEG-4 Audio is under development (as of September 2009).
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng. While there are many MP2 encoders, TooLAME is well-known and widely used for its particularly high audio quality. It has been unmaintained since 2003, but is directly succeeded by the TwoLAME code fork. The name TooLAME is a play on LAME and Layer II.
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.
Unified Speech and Audio Coding (USAC) is an audio compression format and codec for both music and speech or any mix of speech and audio using very low bit rates between 12 and 64 kbit/s. It was developed by Moving Picture Experts Group (MPEG) and was published as an international standard ISO/IEC 23003-3 and also as an MPEG-4 Audio Object Type in ISO/IEC 14496-3:2009/Amd 3 in 2012.
Fraunhofer FDK AAC is an open-source library for encoding and decoding digital audio in the Advanced Audio Coding (AAC) format. Fraunhofer IIS developed this library for Android 4.1. It supports several Audio Object Types including MPEG-2 and MPEG-4 AAC LC, HE-AAC, HE-AACv2 as well AAC-LD and AAC-ELD for real-time communication. The encoding library supports sample rates up to 96 kHz and up to eight channels.
Enhanced Voice Services (EVS) is a superwideband speech audio coding standard that was developed for VoLTE and VoNR. It offers up to 20 kHz audio bandwidth and has high robustness to delay jitter and packet losses due to its channel aware coding and improved packet loss concealment. It has been developed in 3GPP and is described in 3GPP TS 26.441. The application areas of EVS consist of improved telephony and teleconferencing, audiovisual conferencing services, and streaming audio. Source code of both decoder and encoder in ANSI C is available as 3GPP TS 26.442 and is being updated regularly. Samsung uses the term HD+ when doing a call using EVS.
This is what made it possible for modern MP3 encoders (e.g. LAME) to improve far beyond the original L3enc and dist10 reference implementations.
The licensing program coming to an end is due to the fact that the last patent included in the program expired.The page's later caution about third-party implementation–specific patents is not about LAME's implementation. [ citation needed ]