Comparison of audio coding formats. Streaming media. xHE-AAC.
OpusOpus audio format.opus
Vorbis may produce better audio quality with a smaller file size than alternative codecs such as AAC or MP3. However, this has not been proven conclusively. Ogg Vorbis is not bound by patents and is considered "free software" in the sense that no corporate entity owns the rights to the format. Some people feel that this is a safer container for their multimedia content for this reason. However, oggcasters can generally not reach as wide of an audience as more traditional podcasters. This is mainly due to the lack of native Ogg Vorbis support in Microsoft's Internet Explorer and Apple's Safari web browser, and the lack of Ogg Vorbis support in many mobile audio devices.
In addition to the direct applications (MP3 players or computers), digitally compressed audio streams are used in most video DVDs, digital television, streaming media on the internet, satellite and cable radio, and increasingly in terrestrial radio broadcasts. Lossy compression typically achieves far greater compression than lossless compression (data of 5 percent to 20 percent of the original stream, rather than 50 percent to 60 percent), by discarding less-critical data. The innovation of lossy audio compression was to use psychoacoustics to recognize that not all data in an audio stream can be perceived by the human auditory system.
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.
lossylossy data compressioncompressed
Data files using lossy compression are smaller in size and thus cost less to store and to transmit over the Internet, a crucial consideration for streaming video services such as Netflix and streaming audio services such as Spotify. A study conducted by the Audio Engineering Library concluded that lossy compression formats such as MP3s have distinct effects on timbral and emotional characteristics, tending to strengthen negative emotional qualities and weaken positive ones. The study further noted that the trumpet is the instrument most affected by compression, while the horn is least.
ALAC.m4aApple Lossless Audio Codec
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.
Comparison of (audio/video) container formatscontainer files
Comparison of audio coding formats. Enhanced podcast.
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.
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.
Comparison of audio coding formats. Comparison of video codecs.
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
Older formats such as AVI do not support new codec features like B-frames, VBR audio or VFR video natively. The format may be "hacked" to add support, but this creates compatibility problems. 4) Support for advanced content, such as chapters, subtitles, meta-tags, user-data. 5) Support of streaming media. Comparison of (audio/video) container formats. Open source codecs and containers. FFmpeg, cross-platform, open source, audio and video codec suite/library. List of multimedia (audio/video) codecs. Comparison of video codecs. Comparison of audio coding formats. Archive format. Metafile.
MPEG2DVDH.262 / MPEG-2 Part 2 video
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.
Picard supports these audio file formats: * MusicBrainz Picard entries in the MusicBrainz Blog
media playerdigital audio playersportable media players
Some MP3 players have electromagnet transmitters, as well as receivers. Lots of MP3 players have built-in FM radios, but FM transmitters aren't usually built-in due to liability of transmitter feedback from simultaneous transmission and reception of FM. Also, certain features like Wi-Fi and Bluetooth can interfere with professional-grade communications systems such as aircraft at airports. The name MP4 player is a marketing term for inexpensive portable media players, usually from little known or generic device manufacturers. The name itself is a misnomer, since most MP4 players through 2007 were incompatible with the MPEG-4 Part 14 or the .mp4 container format.
VLClibVLCVLC for iOS
Container formats: ASF, AVI, FLAC, FLV, Fraps, Matroska, MP4, MPJPEG, MPEG-2 (ES, MP3), Ogg, PS, PVA, QuickTime File Format, TS, WAV, WebM. Audio coding formats: AAC, AC-3, DV Audio, FLAC, MP3, Speex, Vorbis. Streaming protocols: HTTP, MMS, RTSP, RTP, UDP. Video coding formats: Dirac, DV, H.263, H.264/MPEG-4 AVC, H.265/MPEG-H HEVC, MJPEG, MPEG-1, MPEG-2, MPEG-4 Part 2, Theora, VP5, VP6, VP8, VP9. Comparison of video player software. List of codecs. List of music software.
ID3 is a metadata container most often used in conjunction with the MP3 audio file format. It allows information such as the title, artist, album, track number, and other information about the file to be stored in the file itself. ID3 is also specified by Apple as a timed metadata in HTTP Live Streaming, carried as a PID in the main transport stream or in separate audio TS. There are two unrelated versions of ID3: ID3v1 and ID3v2. ID3v1 takes the form of a 128-byte segment at the end of an MP3 file containing a fixed set of data fields. ID3v1.1 is a slight modification which adds a "track number" field at the expense of a slight shortening of the "comment" field.
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. HE-AAC version 1 profile (HE-AAC v1) uses spectral band replication (SBR) to enhance the compression efficiency in the frequency domain. HE-AAC version 2 profile (HE-AAC v2) couples SBR with Parametric Stereo (PS) to enhance the compression efficiency of stereo signals. It is a standardized and improved version of the AACplus codec.
MPEG-4 is a method of defining compression of audio and visual (AV) digital data. It was introduced in late 1998 and designated a standard for a group of audio and video coding formats and related technology agreed upon by the ISO/IEC Moving Picture Experts Group (MPEG) (ISO/IEC JTC1/SC29/WG11) under the formal standard ISO/IEC 14496 – Coding of audio-visual objects. Uses of MPEG-4 include compression of AV data for web (streaming media) and CD distribution, voice (telephone, videophone) and broadcast television applications.
.flacFree Lossless Audio CodecFLAC
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. Development was started in 2000 by Josh Coalson.
Audio CDCDRed Book
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.
The test sample was a 48 kHz, 5.1 channel surround audio track. Windows Media Video – a video file format and codec developed by Microsoft. JPEG XR / HD Photo – an image file format and codec developed by Microsoft. Surround sound. Timeline of audio formats. Comparison of audio coding formats.
Audible audio files are compatible with hundreds of audio players, PDAs, mobile phones and streaming media devices. Devices that do not have AudibleAir capability (allowing users to download content from their library directly into their devices) require a Windows PC or Macintosh to download the files. Additionally, titles can be played on the PC (using iTunes, Windows Media Player, or AudibleManager). Titles cannot be burned to CD with AudibleManager. According to Audible's website, they can be burned to CD using Apple's iTunes and some versions of Nero.
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: Given that the ear will not be at peak perceptive capacity when dealing with these limitations, a compression algorithm can assign a lower priority to sounds outside the range of human hearing.
sampling ratesamplingsample rate
When it is necessary to capture audio covering the entire 20–20,000 Hz range of human hearing, such as when recording music or many types of acoustic events, audio waveforms are typically sampled at 44.1 kHz (CD), 48 kHz, 88.2 kHz, or 96 kHz. The approximately double-rate requirement is a consequence of the Nyquist theorem. Sampling rates higher than about 50 kHz to 60 kHz cannot supply more usable information for human listeners. Early professional audio equipment manufacturers chose sampling rates in the region of 50 kHz for this reason.