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.

Lossy compression

lossylossy data compressioncompressed
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).


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.

Apple Lossless

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.

Streaming media

streamingstreamedstreaming video
The results of this calculation are as follows: number of MBs transferred = 500 x 1024 (bit/s) × 3 × 3,600 ( = 3 hours) × 3,000 (number of viewers) / (8*1024*1024) = 1,977,539 MB The audio stream is compressed to make the file size smaller using an audio coding format such as MP3, Vorbis, AAC or Opus. The video stream is compressed using a video coding format to make the file size smaller. Video coding formats include H.264, HEVC, VP8 or VP9. Encoded audio and video streams are assembled in a container "bitstream" such as MP4, FLV, WebM, ASF or ISMA.

MPEG-1 Audio Layer II

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.

Data compression

compressionvideo 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).


Ogg VorbisOGGaoTuV
However, trained listeners can often hear significant differences between codecs at identical bitrates, and aoTuV Vorbis performed better than LC-AAC, MP3, and MPC. 2005, July comparison: AAC vs MP3 vs Vorbis vs WMA at 80 kbit/s. States that Vorbis aoTuV beta 4 is the best encoder for either classical or various music in this bitrate, and that its quality is comparable to the LAME ABR MP3 at 128 kbit/s. 2005, August comparison: AAC vs MP3 vs Vorbis vs WMA at 96 kbit/s.


Part 6: Describes extensions for DSM-CC (Digital Storage Media Command and Control). Part 7: Advanced Audio Coding (AAC). Part 8: 10-bit video extension. Primary application was studio video, allowing artifact-free processing without giving up compression. Part 8 has been withdrawn due to lack of interest by industry. Part 9: Extension for real time interfaces. Part 10: Conformance extensions for DSM-CC. Part 11: Intellectual property management (IPMP). An audio compression system limited to two channels (stereo). No standardized support for interlaced video with poor compression when used for interlaced video.


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 coding formats

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.

Discrete cosine transform

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.

Compact Disc Digital Audio

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.

Opus (audio format)

OpusOpus audio format.opus
Comparison of audio coding formats. Streaming media. xHE-AAC. Opus on Hydrogenaudio Knowledgebase.


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.


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.

MP3 player

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.

Digital Audio Broadcasting

DABDAB+DAB Digital Radio
Digital Audio Broadcasting (DAB) is a digital radio standard for broadcasting digital audio radio services, used in many countries around the world, though not North America. The DAB standard was initiated as a European research project in the 1980s. The Norwegian Broadcasting Corporation (NRK) launched the first DAB channel in the world on 1 June 1995 (NRK Klassisk), and the BBC and Swedish Radio (SR) launched their first DAB digital radio broadcasts in 27 September 1995. DAB receivers have been available in many countries since the end of the 1990s.

Windows Media 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.

Audio bit depth

24-bitbit depthresolution
Bits are the basic unit of data used in computing and digital communications. Bit rate refers to the amount of data, specifically bits, transmitted or received per second. In MP3, Ogg and other compressed file format, bit rate is used to encode the number of bits to be transmitted into the particular audio aspect. It is usually measured in kb/s. * Audio system measurements. Color depth—corresponding concept for digital images. Effective number of bits.

Portable media player

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.

Dolby Digital

Dolby Digital 5.1AC3AC-3
Dolby Digital, also known as Dolby AC-3, is the name for audio compression technologies developed by Dolby Laboratories. Originally named Dolby Stereo Digital until 1994, except for Dolby TrueHD, the audio compression is lossy, based on the modified discrete cosine transform (MDCT) algorithm. The first use of Dolby Digital was to provide digital sound in cinemas from 35mm film prints; today, it is now also used for other applications such as TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles.

Code-excited linear prediction

CELPCode Excited Linear PredictionCode-excited linear prediction (CELP)
MPEG-4 Part 3 (CELP as an MPEG-4 Audio Object Type). G.728 – Coding of speech at 16 kbit/s using low-delay code excited linear prediction. G.718 – uses CELP for the lower two layers for the band (50–6400 Hz) in a two-stage coding structure. G.729.1 – uses CELP coding for the lower band (50–4000 Hz) in a three-stage coding structure. Comparison of audio coding formats. CELT is a related audio codec that borrows some ideas from CELP. B.S. Atal, "The History of Linear Prediction," IEEE Signal Processing Magazine, vol. 23, no. 2, March 2006, pp. 154–161. M. R. Schroeder and B. S.

Modified discrete cosine transform

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.


Comparison of audio coding formats. Comparison of video codecs.