Dolby Digital (AC-3). Adaptive Transform Acoustic Coding (ATRAC). MPEG Layer III (MP3). Advanced Audio Coding (AAC / MP4 Audio). Ogg Vorbis (noted for its lack of patent restrictions). Windows Media Audio (WMA) (lossless codec available too). LDAC. Constrained Energy Lapped Transform (CELT). Opus (mostly for real-time applications). Adaptive differential pulse-code modulation (ADPCM). Master Quality Authenticated (MQA). MPEG-1 Audio Layer II (MP2). Musepack (based on Musicam). aptX/ aptX-HD. Linear predictive coding (LPC). Adaptive predictive coding (APC). Code-excited linear prediction (CELP). Algebraic code-excited linear prediction (ACELP). Relaxed code-excited linear prediction (RCELP).
lossylossy data compressioncompressed
Perceptual coding is used by modern audio compression formats such as MP3 and AAC. Discrete cosine transform (DCT), developed by Nasir Ahmed, T. Natarajan and K. R. Rao in 1974, provided the basis for the modified discrete cosine transform (MDCT) used by modern audio compression formats such as MP3 and AAC. MDCT was proposed by J. P. Princen, A. W. Johnson and A. B. Bradley in 1987, following earlier work by Princen and Bradley in 1986. The MDCT is used by modern audio compression formats such as Dolby Digital, MP3, and Advanced Audio Coding (AAC).
DCTiDCTinverse discrete cosine transform
It is used in most digital media, including digital images (such as JPEG and HEIF, where small high-frequency components can be discarded), digital video (such as MPEG, H.26x and Vorbis), digital audio (such as Dolby Digital, MP3 and AAC), digital television (such as SDTV, HDTV and VOD), digital radio (such as AAC+ and DAB+), and speech coding (such as AAC-LD, Siren and Opus). DCTs are also important to numerous other applications in science and engineering, such as digital signal processing, communications devices, reducing network bandwidth usage, and spectral methods for the numerical solution of partial differential equations.
OpusOpus audio format.opus
Comparison of audio coding formats. Streaming media. xHE-AAC. Opus on Hydrogenaudio Knowledgebase.
A podcast is an episodic series of digital audio files that a user can download in order to listen. Alternatively, the word "podcast" may refer to the individual component of such a series or to an individual media file. Podcasting often uses a subscription model, whereby new episodes automatically download via web syndication to a user's own local computer, mobile application, or portable media player. Ben Hammersley originally suggested the word "podcast" as a portmanteau of "iPod" (a brand of media player) and "broadcast" in 2004. The files distributed are in audio format, but may sometimes include other file formats such as PDF or EPUB.
Ultra HD Blu-ray players support decoding of 4K resolution VP8/VP9 video with Ogg Vorbis audio. Comparison of audio coding formats. FreeCast, peer-to-peer Vorbis streaming. Icecast, streaming media server which currently supports Ogg (Vorbis and Theora), Opus and WebM streams. JUCE, cross-platform C++ toolkit with embedded Vorbis support. Ogg bitstream format. Opus, a new audio format by Xiph that may replace Vorbis. Vorbis comment, metadata format used by Vorbis. XSPF, playlist format. Xiph QuickTime Components, official QuickTime implementation. Vorbis site - Xiph reference implementation. Vorbis reference implementation by Xiph.Org Foundation. Players.
ALACApple Lossless Audio Codec.m4a
Apple Lossless, also known as Apple Lossless Audio Codec (ALAC), 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. Apple Lossless supports up to 8 channels of audio at 16, 20, 24 and 32 bit depth with a maximum sample rate of 384kHz.
MP2MPEG-1 Layer IIMUSICAM
This unique characteristic of the MPEG-1 Audio family implies a very good sound quality on audio signals with rapid energy changes, such as percussive sounds. Because both the MP2 and MP3 formats use the same basic sub-band filter bank, both benefit from this characteristic. MPEG-1. MPEG-1 Audio Layer I. MPEG-1 Audio Layer III. MPEG-2. MP4 (container format). Elementary stream. Musepack originally MP2-based, with numerous improvements. Genesis of the MP3 Audio Coding Standard by Hans Georg Musmann in IEEE Transactions on Consumer Electronics, Vol. 52, Nr. 3, pp. 1043–1049, August 2006.
MPEG-4 absorbs many of the features of MPEG-1 and MPEG-2 and other related standards, adding new features such as (extended) VRML support for 3D rendering, object-oriented composite files (including audio, video and VRML objects), support for externally specified Digital Rights Management and various types of interactivity. AAC (Advanced Audio Coding) was standardized as an adjunct to MPEG-2 (as Part 7) before MPEG-4 was issued. MPEG-4 is still an evolving standard and is divided into a number of parts. Companies promoting MPEG-4 compatibility do not always clearly state which "part" level compatibility they are referring to.
MDCTModulated Lapped Transformtime-domain aliasing cancellation
As a result of these advantages, the MDCT is the most widely used lossy compression technique in audio data compression. It is employed in most modern audio coding standards, including MP3, Dolby Digital (AC-3), Vorbis (Ogg), Windows Media Audio (WMA), ATRAC, Cook, Advanced Audio Coding (AAC), LDAC, Dolby AC-4, MPEG-H 3D Audio, as well as speech coding standards such as AAC-LD (LD-MDCT), G.722.1, G.729.1, CELT, and Opus. The discrete cosine transform (DCT) was first proposed by Nasir Ahmed in 1972, and demonstrated by Ahmed with T. Natarajan and K. R. Rao in 1974. The MDCT was later proposed by John P. Princen, A.W. Johnson and Alan B.
MPEG-2 AAC. multichannel encoding with up to 48 channels. Part 1: Systems – describes synchronization and multiplexing of video and audio. (It is also known as ITU-T Rec. H.222.0. ) See MPEG transport stream and MPEG program stream. Part 2: Video – video coding format for interlaced and non-interlaced video signals (Also known as ITU-T Rec. H.262). Part 3: Audio – audio coding format for perceptual coding of audio signals. A multichannel-enabled extension and extension of bit rates and sample rates for MPEG-1 Audio Layer I, II and III of MPEG-1 audio. Part 4: Describes procedures for testing compliance. Part 5: Describes systems for Software simulation.
Free Lossless Audio Codec.flacFLAC
FLAC (Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, 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 decompress to an identical copy of the original audio data. FLAC is an open format with royalty-free licensing and a reference implementation which is free software. FLAC has support for metadata tagging, album cover art, and fast seeking. FLAC's development started in 2000 by Josh Coalson.
Comparison of audio formatsaudio
The following tables compare general and technical information for a variety of audio coding formats. For listening tests comparing the perceived audio quality of audio formats and codecs, see the article Codec listening test. 1) The 'Music' category is merely a guideline on commercialized uses of a particular format, not a technical assessment of its capabilities.
Comparison of audio coding formats. Comparison of video codecs.
Common audio file formats for this purpose include WAV and AIFF, which simply preface the LPCM data with a short header; FLAC, ALAC, and Windows Media Audio Lossless, which compress the LPCM data in ways that conserve space yet allow it to be restored without any changes; and various lossy, perceptual coding formats like MP3 and AAC, which modify and compress the audio data in ways that irreversibly change the audio, but that exploit features of human hearing to make the changes difficult to discern. Recording publishers have created CDs that violate the Red Book standard. Some do so for the purpose of copy prevention, using systems like Copy Control.
Pulse-code modulation (PCM) is a method used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Linear pulse-code modulation (LPCM) is a specific type of PCM where the quantization levels are linearly uniform. This is in contrast to PCM encodings where quantization levels vary as a function of amplitude (as with the A-law algorithm or the μ-law algorithm).
Nasir AhmedN. Ahmed
The DCT has been widely cited in patents that have been awarded since 1976, as evident from the following results corresponding to various search scenarios: A DCT variant, the modified discrete cosine transform (MDCT), is used in modern audio compression formats such as MP3, Advanced Audio Coding (AAC), and Vorbis (OGG). The discrete sine transform (DST) is derived from the DCT, by replacing the Neumann condition at x=0 with a Dirichlet condition. The DST was described in the 1974 DCT paper by Ahmed, Natarajan and Rao. Ahmed later was involved in the development a DCT lossless compression algorithm with Giridhar Mandyam and Neeraj Magotra at the University of New Mexico in 1995.
psychoacousticpsychoacoustic modelperceptual coding
Such compression is a feature of nearly all modern lossy audio compression formats. Some of these formats include Dolby Digital (AC-3), MP3, Opus, Ogg Vorbis, AAC, WMA, MPEG-1 Layer II (used for digital audio broadcasting in several countries) and ATRAC, the compression used in MiniDisc and some Walkman models. Psychoacoustics is based heavily on human anatomy, especially the ear's limitations in perceiving sound as outlined previously. To summarize, these limitations are: A compression algorithm can assign a lower priority to sounds outside the range of human hearing.
meta datameta-datacommunications metadata
As different digital audio formats were developed, attempts were made to standardize a specific location within the digital files where this information could be stored. As a result, almost all digital audio formats, including mp3, broadcast wav and AIFF files, have similar standardized locations that can be populated with metadata. The metadata for compressed and uncompressed digital music is often encoded in the ID3 tag. Common editors such as TagLib support MP3, Ogg Vorbis, FLAC, MPC, Speex, WavPack TrueAudio, WAV, AIFF, MP4, and ASF file formats.
container formatcontainercontainer formats
MPEG program stream (standard container for MPEG-1 and MPEG-2 elementary streams on reasonably reliable media such as disks; used also on DVD-Video discs). MPEG-2 transport stream (a.k.a. MPEG-TS) (standard container for digital broadcasting and for transportation over unreliable media; used also on Blu-ray Disc video; typically contains multiple video and audio streams, and an electronic program guide). MP4 (standard audio and video container for the MPEG-4 multimedia portfolio, based on the ISO base media file format defined in MPEG-4 Part 12 and JPEG 2000 Part 12) which in turn was based on the QuickTime file format.
Picard supports these audio file formats: * MusicBrainz Picard entries in the MusicBrainz Blog MusicBrainz Picard entries in the MusicBrainz Blog.
digital audio playerMP3 playersmusic player
An MP3 player is an electronic device that can play MP3 digital audio files. It is a type of digital audio player, or portable media player. Most players play more than the MP3 file format, such as Windows Media Audio (WMA), Advanced Audio Coding (AAC), Vorbis, FLAC, Speex and Ogg. In 1981, Kane Kramer filed for a UK patent for the IXI, the first Digital Audio Player. UK patent 2115996 was issued in 1985, and U.S. Patent 4,667,088 was issued in 1987. The player was as big as a credit card and had a small LCD screen, navigation and volume buttons and would have held at least 8MB of data in a solid-state bubble memory chip with a capacity of 3½ minutes' worth of audio.
WMAWMA LosslessWMA Pro
However, a September 2003 public listening test conducted by Roberto Amorim found that listeners preferred 128 kbit/s MP3 to 64 kbit/s WMA audio with greater than 99% confidence. At 80 kbit/s and 96 kbit/s, WMA had lower quality than HE-AAC, AAC-LC, and Vorbis; near-equivalent quality to MP3, and better quality than MPC in individual tests done in 2005. At 128 kbit/s, there was a four-way tie between aoTuV Vorbis, LAME MP3, WMA 9 Pro and AAC in a large scale test in January 2006, with each codec sounding close to the uncompressed music file for most listeners.
media playerportable media playersdigital audio players
Nearly all players are compatible with the MP3 audio format, and many others support Windows Media Audio (WMA), Advanced Audio Coding (AAC) and WAV. Some players are compatible with open-source formats like Ogg Vorbis and the Free Lossless Audio Codec (FLAC). Audio files purchased from online stores may include digital rights management (DRM) copy protection, which many modern players support. The JPEG format is widely supported by players. Some players, like the iPod series, provide compatibility to display additional file formats like GIF, PNG, and TIFF, while others are bundled with conversion software.
MPEGMotion Picture Experts GroupMPEG encoding
Audio codec. Audio coding format. Discrete cosine transform (DCT). Video codec. Video coding format. Video quality. Video compression. MP3. Gary Sullivan (engineer). Hiroshi Yasuda. Nasir Ahmed (N. Ahmed). Leonardo Chiariglione. Official MPEG web site. MPEG.ORG. Papers and books on MPEG.