Hydrogenaudio Knowledgebase - User contributions [en]

Foobar2000:Foobar2000

2026-01-15T23:16:27Z

Logan589XP: Updated stable release version number and release date.

{{Software Infobox|
|name = foobar2000
|logo = [[Image:foobar2000 Logo.png|48px]]
|screenshot = [[Image:Foobar2000-1.0-default-ui.png|250px]]
|caption = Screenshot of foobar2000 v1.0 using the default user interface
|developer = Peter Pawlowski
|released = {{start date and age|2002|12|20}}<ref>{{ha|https://hydrogenaud.io/index.php/topic,4997.0.html|Official Foobar2000 site & Foobar2000 0.3 & SDK!}}</ref>
|stable_release = 2.24.5<ref>{{foobar2000|https://www.foobar2000.org/download|Download foobar2000 for Windows}}</ref>
|stable_release_date = 2026-01-02
|preview_release =
|preview_release_date =
|operating_system = Windows
|use = Media Player
|license = Proprietary, BSD
|website = [https://www.foobar2000.org/ foobar2000.org]
}}
'''foobar2000''' is an advanced freeware audio player for the Windows platform. Some of the basic features include full Unicode support, [[ReplayGain]] support and native support for several popular audio formats.

== Platforms ==

foobar2000 has been written specifically for the Windows platform, and there are no plans to port it to any others. However, although not officially supported, it is known to run on [https://www.hydrogenaudio.org/forums/index.php?showtopic=54933 Linux] and [https://www.hydrogenaudio.org/forums/index.php?showtopic=77261 Mac OS X] through Wine and WineBottler, respectively. Also there is an [https://aur.archlinux.org/packages/foobar2000 Arch Linux native package] maintained by a community member.

== Features ==
* Powerful open component architecture allowing third-party developers to extend functionality of the player, including the ability to fully replace the user interface.
* Full Unicode support: File names, user interface, tagging, etc.
* [[ReplayGain]] support: Both playback and writing ReplayGain information to file tags.
* [[Gapless playback]].
* Advanced [[tagging]] capabilities - through built-in [[foobar2000:Properties|Properties dialog]] and various optional tagging-related components.
* Built-in [[foobar2000:Preferences:Media Library|Media Library]] functionality.
** Intuitive [[foobar2000:Query syntax|query syntax]] for searching the Media Library.
** Autoplaylist support: Generate dynamically updating playlists based on queries.
* [[foobar2000:Preferences:General:Keyboard Shortcuts|Customizable keyboard shortcuts]].
* Support for [[transcoding]] all supported audio formats using the [[Foobar2000:Converter|Converter component]] (requires external command-line encoder executables for different output formats).
* [[Secure_ripping|Secure]] [[foobar2000:Ripping CDs|CD ripping]].
* Streaming support.
* Efficient handling of large playlists.
* [[foobar2000:Components/Default_user_interface_%28foo_ui_std%29|User interface]] with simple configuration to create even complex layouts quickly and easily.
* Highly customizable display of track information using [[foobar2000:Titleformat_Introduction|title formatting scripts]].

==Supported audio formats==
Natively supported ("out-of-the-box"):
* [[MP1]], [[MP2]], [[MP3]], [[MP4]], [[Musepack]], [[AAC]], [[Ogg Vorbis]], [[FLAC]] / Ogg FLAC, [[Speex]], [[WavPack]], [[WAV]], [[AIFF]], [[AU|AU/SND]], [[CDDA]], [[WMA]], [[Matroska]], [[ALAC]], [[MMS]], [[RTSP]], [[Opus]], [[Monkey's Audio]], [[AC3]], [[DTS]], [[TAK]].
Supported through optional components:
* [[TTA]], [[MOD]], [[SPC]], [[Shorten]], [[OptimFROG]], [[PSF]], [[NSF]], [[XID]], [[XA]], [[AMR]], ''etc.''

In addition, foobar2000 can also play audio files within ZIP and RAR archives (typically compressed) directly, ''i.e.'' without requiring the user to extract the files first. More archive formats are supported ''via'' additional components: [https://www.foobar2000.org/components/view/foo_unpack_lha LHA/LZH], [https://www.foobar2000.org/components/view/foo_unpack_7z 7-Zip]

== Using foobar2000 ==

=== Important pages ===
* {{foobar2000|https://www.foobar2000.org/FAQ|foobar2000 FAQ}}
* [[foobar2000:components|foobar2000 components]]
* [[foobar2000:Legacy components|foobar2000 legacy components]]
* [[foobar2000:Encouraged Tag Standards|foobar2000 encouraged tag standards]]
* [[foobar2000:FAQ|foobar2000 FAQ (unofficial)]]
* [[:Category:Foobar2000_Guides|foobar2000 guides (category)]]
* [[:Category:Foobar2000_Preferences|foobar2000 preferences (category)]]

=== Specific guides ===

'''Preferences'''
* [[foobar2000:Preferences|Preferences dialog]]

'''Metadata'''
* [[foobar2000:Properties|Tag editing: the Properties dialog]]
* [[foobar2000:Query syntax|Query Syntax]]: details of Syntax for querying metadata.

'''Title formatting'''
* [[foobar2000:Title Formatting Introduction|Introduction to titleformat scripts]]
* [[foobar2000:Title Formatting Reference|Titleformat Reference]]: reference guide to all fields and functions
* [[foobar2000:Titleformat Examples|Titleformat Examples]]: user-submitted code for various purposes; submit your own!

'''Others'''
* [[foobar2000:File operations|File operations dialog]]: move, copy, rename, and delete files from within foobar2000
* [[foobar2000:Commandline Guide|Commandline usage]]

=== External Guides ===
* [https://www.audiohq.de/viewforum.php?id=27 German-language guides]
* [https://foobar2000.xrea.jp/ fb2k Wiki Page] for Japanese users.
* [https://winamp2foobar.blogspot.com Winamp To Foobar Guide] with information relevant for general users also.

=== Technical information ===
* [[foobar2000:ID3 Tag Mapping|ID3 Tag Mapping]]
* [[foobar2000:Metadata Compatibility|Metadata Compatibility]]: information about compatibility with metadata written by other applications

== Important Links ==
=== Official site ===
* [https://www.foobar2000.org foobar2000.org: Homepage]
* [https://www.foobar2000.org/download foobar2000.org: Download]
* [https://www.foobar2000.org/components foobar2000.org: Components]

=== Community ===
* Official IRC Channel: #foobar2000 on [https://libera.chat/ Libera Chat]
* {{ha|https://hydrogenaud.io/index.php/board,28.0.html|foobar2000 Forums}}
* [https://foobar-users.de/ German Support Forum]
* [http://www.fforum.ru/viewforum.php?f=59 Russian-language forum]
* [https://www.foobar2000.ru/forum/ Another Russian-language forum]

=== Appearance ===
* {{ha|https://hydrogenaud.io/index.php/topic,61333.0.html|Default UI .fth thread}} (fast way to clone another's DUI configuration)
* [[foobar2000:Preferences:Columns UI/Appearance|Columns UI appearance customization guides]]
* {{ha|https://hydrogenaud.io/index.php/topic,31027.0.html|Columns UI configurations}}

== References ==
<references/>

[[Category:foobar2000]]
[[Category:Media Players|foobar2000]]
[[Category:CD Rippers]]
[[Category:Software]]
[[Category:Tag editors]]

Opus

2026-01-15T23:14:08Z

Logan589XP: Added release notes for libopus 1.6 and updated stable release number to 1.6.1.

{{Software Infobox
| name = Opus
| logo = [[Image:opus-logo.png|250px|Official Opus logo]]
| screenshot =
| caption = Opus Interactive Audio Codec
| maintainer = [http://xiph.org/ Xiph.Org Foundation]
| stable_release = 1.6.1
| operating_system = Windows, Mac OS/X, Linux/BSD
| use = Encoder/Decoder
| license = 3-clause BSD license
| website = [http://www.opus-codec.org/ opus-codec.org]
}}

'''Opus''' is a [[lossy]] audio compression format developed by the Internet Engineering Task Force (IETF) designed to be suitable for interactive real-time applications over the Internet,{{ref|homepage|a}} including music as well as speech, yet it is also very competitive for use as a storage and playback format, being a [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ class leader at around 64 kbps] and [http://listening-test.coresv.net/results.htm also at 96 kbps]. As an open format standardised through [http://tools.ietf.org/html/rfc6716 Request for Comments (RFC) 6716],{{ref|RFC|c}} a high quality reference implementation is provided under the 3-clause BSD license{{ref|homepage|a}} which compiles and runs on the vast majority of general purpose and embedded (fixed point) processors. Many Software patents which cover Opus are licensed under royalty-free terms.{{ref|FAQ|b}} Opus is also a Mandatory To Implement (MTI) codec for the upcoming WebRTC (Web Real Time Communication) specification of the World Wide Web Consortium (W3C).

Opus incorporates technology from two codecs, the speech-oriented SILK codec developed by Skype and the multi-purpose low-latency CELT codec developed by Xiph.org with significant changes to each to ensure they can work together.{{ref|RFC|c}} Opus can seamlessly transition among high and low bitrates, using a linear prediction codec (the SILK layer) at lower bitrates and a lapped transform codec (the CELT layer) at higher bitrates, as well as a hybrid of the two for a short overlap in which SILK encodes the 0–8 kHz spectrum and the CELT layer encodes only the frequencies above 8kHz.{{ref|RFC|c}} Opus has very low algorithmic delay (typ 22.5 ms) compared to popular music formats such as [[MP3]], [[Vorbis |Ogg Vorbis]], [[AAC | LC-AAC and HE-AAC]] (all over 100 ms), yet performs very competitively with them in terms of quality per bitrate, making it comparably viable as a storage & playback format. Also unlike Vorbis, Opus does not require the definition of large codebooks for each individual file, making it also preferable for short clips of audio, such as those often used by game developers, a field where patent-free Vorbis is commonly used.{{ref|RFC|c}}

Considerably more details of the history and potential applications for Opus are included in the ''Wikipedia'' page for '''[http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Opus (audio format)]'''

==Characteristics==
Opus supports bitrates from 6 kbps to 510 kbps for typical stereo audio sources (and a maximum of around 255 kbps per channel for multichannel audio), with the 'sweet spot' for music and general audio around 30 kbps (mono) and 40–100 kbps (stereo). It is intrinsically [[VBR | variable bitrate]], though constrained VBR and [[CBR | constant bitrate]] modes are possible where required. In the case of the reference release, libopus, the target bitrate is calibrated against the internal constant quality targets so that over a typical music collection, something very close to the target bitrate will be achieved. This bitrate-calibrated approach differs from most VBR encoders (e.g. LAME, helix mp3, qaac, Nero aacenc, Ogg Vorbis, Musepack) where a setting on some 'constant quality' scale (which differs between encoders) is used and the bitrate will fall where it may. Improved future versions can be expected to offer improved quality at the same setting. Independent implementations may adopt a different approach.

Opus is able to seamlessly adapt its mode of operation without glitches or sound interruption (an illustrative demonstration of [http://opus-codec.org/examples/#gauge bitrate scalability] is on the Opus Examples page), which can be particularly useful for mixed-content audio or varying network conditions, making the unified Opus codec superior to a suite of different codecs that might otherwise cover the same range of bitrate and quality settings and would require out-of-band signalling to instigate codec switching. The switching includes the choice of mono, stereo and other channel mappings, the use of the speech-oriented SILK layer, the general-purpose CELT layer or the hybrid of both, and the use of different audio bandwidths (4, 6, 8, 12, or 20 kHz) as well as the quality adjustments within the same operating mode that are available in most VBR-capable codecs.

Of importance mainly to interactive uses, but potentially useful in time-delayed audio streaming also, Opus includes packet loss concealment (PLC) in all modes and, in the speech-oriented modes where the SILK layer is active it also supports Forward Error Correction (FEC) where the expected rate of packet loss can be indicated to the encoder by the user or by application software and critical frames (e.g. consonant sounds) can be retransmitted at low bitrate to preserve intelligibility.

For music and general audio, the CELT layer of Opus builds on knowledge gained during xiph.org's Vorbis development and ensures as a primary goal that the total energy in each spectral band is preserved while requiring only a modest bitrate overhead to achieve this, thereby eliminating a lot of bitrate-starvation artifacts such as 'birdies' that are common in low-bitrate MP3, especially during transients, applause and cymbal sounds. This technique likewise increases coding efficiency at bitrates targetting transparent music reproduction. Short blocks (2.5 ms) are also possible for efficient transient handling. Short blocks can also be used exclusively, if very low algorithmic delay (5.0 ms) is required to enable very low-latency interative audio (e.g. live networked music performances such as remote jam sessions), though greater bitrate is then required to maintain the same quality (illustrated in [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo Monty's CELT demo page] under Constant PEAQ value, varying latency). CELT uses a number of additional techniques and provides additional advanced tools to enable encoder tuning.

Opus natively supports [[gapless playback]] (though [[Gapless_playback#Poorly_designed_playback_systems | poor player design]] might itself induce interruptions during playback). Playback gain is also required, making some form of [[ReplayGain]] or [[ReplayGain_2.0_specification | similar]] volume control possible in any compliant player.

==Bitrate performance==
For mono speech, Opus ranges from intelligible narrowband speech reproduction starting at 6 kbps to medium-band, wideband and superwideband speech, reaching full-band speech by around 14 kbps in encoder version 1.2 (was 21 kbps in v1.1, 29 kbps in v1.0). Above about 32 kbps, the SILK layer is no longer used at all, as CELT alone gives superior quality.

For music, the SILK modes are quite tolerable and better than CELT at very low bitrates. The hybrid mode is adopted as bitrate increases, extending bandwidth first to 12 kHz (comparable with compact cassette) then to the full 20 kHz and CELT then takes over. Assuming the source is stereo, the transition from mono to stereo typically happens between the transition from 12 kHz to 20 kHz. Encoder version 1.2 includes great improvements to music encoding in the 32–64 kbps range, allowing full-band stereo at 32 kbps and providing acceptable quality at 48 kbps where artifacts are audible but rarely annoying. Version 1.3 is expected to further improve quality in this range.

Multi-format stereo music listening tests have demonstrated the superiority of Opus at 64 kbps and 96 kbps compared to the best AAC-LC, HE-AAC and Ogg Vorbis encoders, and at 96 kbps also to 128 kbps MP3 encoded using LAME <code>-V 5</code>.

==Indicative bitrate and quality==
The tables below give illustrative, indicative quality guidance based on typical modes used internally by Opus and a range of listening tests.

In encoder version 1.1 automatic detection of speech/music and bandwidth detection were introduced to improve mode decisions and VBR is less constrained, all with the aim of maximizing the quality/bitrate tradeoff, and these improvements are further enhanced in version 1.2 and 1.3. These tables are likely to require updates as the encoder is improved, especially in low-bitrate regions.

===Speech encoding quality===
This table assumes a '''monophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate) but mentions stereo compatibility for 40kbps+. The default 20ms frame size (22.5ms latency) is assumed. Note that the selection of ''VOIP'' mode will deliberately modify the sound with a High Pass Filter and emphasis of formants and harmonics to improve intelligibility of speech especially in noisy environments much as telephones do. ''Auto'' mode will not modify the sound prior to encoding so is usually better for high quality speech recordings or mixed speech and music.

{| class="wikitable" style="text-align:center"
|-
!Bitrate Target
!Bandwidth
!Typical Mode Used
!Speech Quality
!Use Cases / Competitive Codecs
|-
! Less than 6 kbps
| —
| —
| Bitrates lower than 6 kbps not supported by Opus (SILK disabled if forced to encode, which results in terrible speech quality)
| Try [https://en.wikipedia.org/wiki/Codec_2 Codec 2] for 0.45–3.2 kbps mono speech or [[Wikipedia:Lyra (codec)|Lyra]] for 3.2 kbps mono speech
|-
!6 kbps
|4 kHz narrow-band
|SILK
|Fair, intelligible
|AMR-NB may be a little better, but higher latency & proprietary, [[Speex]] also competitive
|-
!9 kbps VBR/CVBR 10 kbps CBR
|8 kHz wide-band
|SILK
|Telephone quality
|AMR-NB & AMR-WB similar quality, but higher latency & proprietary. [[Speex]] competitive.
|-
!12 kbps
|12 kHz super-wideband
|hybrid
|Medium bandwidth, better than telephone quality
|Similar quality to AMR-WB
|-
!16 kbps
|20 kHz
|hybrid/CELT
|Wideband speech quality
|Similar to/better than AMR-WB
|-
!24 kbps
|20 kHz
|hybrid/CELT
|Near transparent speech
|Better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!32 kbps
|20 kHz
|CELT
|Essentially transparent speech plus moderately good stereo music
|Much better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!40 kbps
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, fairly good stereo music
|Stereo podcasts/audiobooks/talk radio with some music
|-
!48 kbps or more
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, reasonable music
|Flexible general purpose modes to suit mixed music and speech
|-
|}

One major limitation of Opus at low bitrate is that SILK is inherently VBR: it accepts no constraints in CVBR, and if forced to do CBR the quality degrades from bit-shaving. As a result, even though constrained VBR is designed such that a fixed-rate data link requires at most one frame of buffer to handle the variation in bit rate -- great news for communication links -- any use of SILK, even in hybrid mode, has the potential of breaking this intention. This makes Opus suboptimal for low-rate radio links: radio links requires a predictable buffer amount, which is only possible with CBR when SILK is used, but use of CBR in turn hurts SILK. There is a noticeable quality difference at the NB/WB switch at 9 kbps VBR / 10 kbps CBR.

Opus 1.3+ allows forced use of SILK down to 5 kbps VBR (NB) and 6 kbps VBR (WB, requires forcing the C API with <code>OPUS_SET_BANDWIDTH</code>). However, quality is in no way guaranteed -- it's just possible.

===Music encoding quality===
This table assumes a '''stereophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate). Opus will automatically use mono at very low bitrates, though a certain amount of stereo encoding can still be used (content dependent) even when mono is specified as the typical stereo mode in the table below.

{| class="wikitable" style="text-align:center"
|-
!Bitrate target
!Stereo mode
!Bandwidth
!typ SILK/CELT use
!Music quality notes
!Use cases/notes/competitive codecs
|-
!4 kbps
|mono
|6 kHz
|SILK
|Poor, muffled sound but intelligible lyrics.
| —
|-
!9 kbps
|mono
|8 kHz
|SILK
|Poor, muffled but OK for bitrate
| —
|-
!14 to 16 kbps
|mono
|20 kHz
|hybrid/CELT
|Fairly poor but OK for bitrate
|Perhaps acceptable for incidental music
|-
!22 to 24 kbps
|mono
|20 kHz
|hybrid/CELT
|Fair but OK for bitrate
|OK for incidental music
|-
!32 to 40 kbps
|stereo
|20 kHz
|CELT
|Moderately good stereo, some artifacts, rarely nasty
|Stereo podcasts, audiobooks, very low bitrate music
|-
!48 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, may have problems with cymbals
|Stereo podcasts, audiobooks, low bitrate music
|-
!64 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, detectable differences to original (mostly 'not annoying')
|Music storage & streaming. Beat HE-AAC, Vorbis, MP3 in [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ listening test]
|-
!96 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, good quality approaching transparency
|Music storage & high quality streaming. Beat LC-AAC, Vorbis, MP3 in [http://listening-test.coresv.net/results.htm listening test]
|-
!112 kbps
|stereo
|20 kHz
|CELT
|Fairly close to transparency (needs more testing)
|Music storage & high quality streaming. Very low-latency stereo networked music performance/jam sessions at OK quality (see below table)
|-
!128 kbps
|stereo
|20 kHz
|CELT
|Very close to transparency (needs more testing). Most modern codecs competitive (AAC-LC, Vorbis, MP3)
|Music storage & streaming. Future download music sales.
|-
!160 to 192 kbps
|stereo
|20 kHz
|CELT
|Transparent with very low chance of artifacts (a few killer samples still detectable). Most old & new lossy codecs competitive.
|Music storage & streaming, dedicated limited-bandwidth audio links (e.g. wireless, [http://en.wikipedia.org/wiki/Bluetooth_profile#Advanced_Audio_Distribution_Profile_.28A2DP.29 A2DP-bluetooth] type links).
|-
!510 kbps
|stereo
|20 kHz
|CELT
|Maximum possible stereo bitrate target (actual rate often less than 510 for default frame size). Most old and new lossy codecs competitive, plus near-lossless [[lossyWAV]] and [[WavPack | WavPack lossy]]
|Music storage, dedicated limited-bitrate audio links (e.g. wireless, minimum latency high quality audio. LossyWAV and WavPack lossy are very competitive for storage, and WavPack lossy <code>--blocksize=256</code> may be competitive with minimum latency mode also.
|-
!>510 kbps
| —
| —
| —
|Above Opus bitrate range allowed for stereo sources
|Settle for 510 kbps or use [[lossless]], [[lossyWAV]], [[WavPack | WavPack lossy]] or lossy transform/subband codecs like [[Vorbis]], [[Musepack]] at very high settings.
|-
|}

===Lower latency versus quality/bitrate trade-off===
====Packet overhead in interactive applications====
For interactive use on the Internet or other packet-based networks, total bandwidth used will be subject to packet overhead. The more packet headers that are transmitted every second, the greater will be the overhead that is required. For this reason, Opus, while defaulting to 20 ms frames, supports 60 ms frames to reduce overhead when transporting low-bitrate SILK frames at the expense of greater latency, which may still be acceptable for speech, and also supports 10 ms SILK frames to reduce latency somewhat at the expense of packet overhead.

In the CELT layer, which tends to operate at higher bitrates than SILK, 20 ms frames are the default, but frames of 10 ms, 5 ms and 2.5 ms are also possible, which directly increases the frame overhead by transmitting more packets per second to achieve lower latency. In addition, as we'll see below it also reduces the quality/bitrate tradeoff of the CELT layer itself.

You probably do not want to use a frame size lower than 10 ms in applications containing speech, as doing so turns off SILK. The "lowdelay" application switch (available in FFmpeg and the raw library) turns off SILK to cut out 4 ms of synchronization delay, but a frame size of 10 ms achieves more delay reduction compared to default without sacrificing SILK.

None of the bitrates mentioned in this article account for the packet overhead.

====CELT layer latency versus quality/bitrate trade-off====
Unlike the SILK layer, which works on fixed 10 ms blocks, 1, 2 or 6 of which can be combined into an Opus frame, the CELT layer is able to modify the encoding block lengths available to enable its use with shorter frames.

When the CELT layer uses 10 ms, 5 ms and 2.5 ms frames instead of the default 20 ms, it must use smaller transform block sizes to achieve this, thereby reducing frequency resolution in the MDCT compared to the default transform window, thus reducing encoding efficiency for tonal signals. To obtain the same frequency precision for a sound divided into shorter transform windows, improved amplitude precision is necessary, resulting in increased bitrate to obtain the same perceptual quality (or conversely lower quality at the same bitrate).

These reduced-latency modes remain efficient for transient signals, which use short blocks anyway.

In all modes, the algorithmic delay consists of the frame size plus an additional 2.5 ms delay. The CELT layer requires 2.5 ms for MDCT window overlap.

Xiph.org used matched [[PEAQ]] scores (approximate perceptual quality assessment made in software) for the CELT0.10 codec that was used as the basis of the CELT layer in the Opus reference release, which indicate the following [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo approximate equivalent settings] for stereo music.

{| class="wikitable" style="text-align:center"
|-
!Frame size
!Algorithmic delay
!Bitrate to match 64kbps@22.5ms delay
!fractional bitrate increase
|-
!20 ms
|22.5 ms
|64.0 kbps
| +0.0 %
|-
!10 ms
|12.5 ms
|70.4 kbps
| +10.0 %
|-
!5 ms
|7.5 ms
|84.8 kbps
| +32.5 %
|-
!2.5 ms
|5.0 ms
|112.0 kbps
| +75.0 %
|-
|}

N.B. This table is useful for interactive streaming only. For music storage & delayed playback or non-interactive streaming, latency reduction is not important and the default 20 ms frame size is preferable.

==== "Equivalent bitrate" ====
Opus code includes a [https://github.com/xiph/opus/blob/9fc8fc4cf432640f284113ba502ee027268b0d9f/src/opus_encoder.c#L806 {{code|compute_equiv_rate()}}] function. Given the bitrate, framesize, cbr decision, and complexity setting, it converts the bitrate to an standard config (VBR, 20 ms frame, complexity 10) equivalent to be used for bandwidth, layer, and stereo decisions. The interesting bits are:

* CBR requires 8% more bitrate for the same quality.
* Frame overhead is fixed and modelled as <code>(40*channels+20)*(frame_rate - 50)</code> for any frame_rate larger than 50. (frame_rate is the number of frames per second, so <code>1000/frame_size_ms</code>). There's no modelling for reduction in overhead from larger-than-standard frames: you'd imagine the expression runs in the opposite direction as well.
* Complexity turning results in up to 30% more bitrate requirement.

This layer of conversion is why Opus runs wideband speech at 9 kbps VBR and CVBR, but with CBR it takes 10 kbps (now we know it's exactly 9.75 kbps) to use WB.

=== Channel count vs bitrate ===

For surround sound bitrates, use [[Bitrate#Equivalent bitrate estimates for multichannel audio]].

For ambisonics, see [https://www.mdpi.com/2076-3417/10/9/3188 AMBIQUAL listening test], paper figures 11 and 12.

== Implementations ==

The format and algorithms are openly documented and the reference implementation is published as free software. The reference implementation (Opus Audio Tools, opus-tools), consisting of separate encoders and decoders, is published under the terms of a BSD-like license. It is written in C programming language and can be compiled for hardware architectures with or without floating point unit. The accompanying diagnostic tool opusinfo reports detailed technical information about Opus files, including information on the standard compliance of the bitstream format. It is based on ogginfo from the vorbis-tools and therefore, unlike the encoder and decoder, available under the terms of version 2 of the GPL.

=== Reference implementation (libopus + binaries) ===
The commandline tools of the reference version are available pre-compiled for the most popular operating systems at [http://opus-codec.org/downloads opus-codec.org] and [https://ftp.mozilla.org/pub/mozilla.org/opus/ Mozilla's ftp server], plus in the foobar2000 free encoders pack and some alternative compiles through the hydrogenaud.io opus forum. The libopus commandline tools include encoder <code>opusenc</code>, decoder <code>opusdec</code>, and with a different license, the <code>opusinfo</code> opus stream & metadata analyzer.

The '''latest stable release''' is recommended for general use and as of mid 2014 is considered competitive with or superior to the best alternative speech or general music encoders at most supported bitrates.

==== libopus v1.0 ====
Released 11 Sep 2012 when RFC6716 was standardized but mostly fully developed by late 2011.

'''Stable, well-tuned''' <code>opusenc</code> reference encoder as included in RFC documentation.

CELT layer closely related to CELT 0.10 implements Constrained VBR mode by default (bitrate boost used mainly for transients), plus true CBR.

==== libopus v1.1 ====

The alpha source code released 21 Dec 2012 for testing & user feedback and following a beta release and testing, the stable 1.1 version was released on 5 December 2013, considered well tested enough for general release.<ref>https://people.xiph.org/~xiphmont/demo/opus/demo3.shtml</ref>

CELT layer [http://jmspeex.livejournal.com/11737.html quality improvements] introduced to provide '''unconstrained VBR''' include a rate boost not just for transients but now for highly tonal signals too and rate reduction when stereo image is narrow. There's also a rewrite of its '''transient detection''' code and '''time-frequency analysis''' code, and rewritten '''dynamic allocation''' code (HF/LF tilt and Band Boost) to allow more aggressive changes from the typical static allocation when warranted.

There are many minor improvements to '''speech quality''' in both SILK and CELT layers.

*'''DC-rejection''' below 3 Hz also aids quality if inaudible DC offset is present with no effect on deep bass notes.
*'''Automatic speech/music detection''' is introduced to optimize encoding mode choices, especially near the bitrate target range (presumably around 24–40 kbps) where the encoder may perform best with SILK, hybrid or CELT depending on content type. Below that range SILK performs best for both music & speech, and above it CELT performs best for speech & music. The detection, without look-ahead is not perfect but usually is undecided in audio where either mode will work well.
*'''Automatic bandwidth detection''' is also introduced to save wasted bits allocated to absent frequencies.
*'''Surround sound improvements''' were introduced since the beta release with considerable advances in coding efficiency, bitrate allocation and quality.

A new '''temporal VBR''' feature is added. For reasons not explained by classic psychoacoustics, it appears that giving more bits to loud frames (stealing from quiet frames) makes the result substantially better on listening tests. This feature is not tunable: it always affects VBR calculation at low bitrates, gradually becoming weaker at higher bitrates, until it turns off completely at 68 kbps.

==== libopus v1.1.3 ====
Released July 15th, 2016. This version contains:

*Neon optimizations improving performance on ARMv7 and ARMv8 by up to 15%
*Fixes some issues with 16-bit platforms (e.g. TI C55x)
*Fixes to comfort noise generation (CNG)
*Documenting that PLC packets can also be 2 bytes
*Includes experimental ambisonics work (<code>--enable-ambisonics</code>)

==== libopus v1.2.1 ====
Released June 26th, 2017. This version contains:

*Speech quality improvements especially in the 12–20 kbit/s range
*Improved VBR encoding for hybrid mode
*More aggressive use of wider speech bandwidth, including fullband speech starting at 14 kbit/s
*Music quality improvements in the 32–48 kbit/s range
*Generic and SSE CELT optimizations
*Support for directly encoding packets up to 120 ms
*DTX support for CELT mode
*SILK CBR improvements
*Support for all of the fixes in draft-ietf-codec-opus-update-06 (the mono downmix and the folding fixes need <code>--enable-update-draft</code>)
*Many bug fixes, including integer wrap-arounds discovered through fuzzing (no security implications)

==== libopus v1.3 ====
Released on October 18th, 2018. This version contains:

* Improvements to voice activity detection (VAD) and speech/music classification using a recurrent neural network (RNN)
* Support for ambisonics coding using channel mapping families 2 and 3
* Improvements to stereo speech coding at low bitrate
* Using wideband encoding down to 9 kb/s
* Making it possible to use SILK down to bitrates around 5 kb/s
* Minor quality improvement on tones
* Enabling the spec fixes in <nowiki>RFC 8251</nowiki> by default
* Security/hardening improvements
* Fixes to the CELT PLC
* Bandwidth detection fixes

==== libopus v1.3.1 ====
Released on April 12th, 2019. This version contains:

* Fixes to x87 builds
* A new OPUS_GET_IN_DTX query to know if the encoder is in DTX mode (last frame was either a comfort noise frame or not encoded at all)
* A new (and still experimental) CMake-based build system that is eventually meant to replace the VS2015 build system (the autotools one will stay)

==== libopus v1.4 ====
Released on April 20th, 2023. This version contains:

* Improved tuning of the Opus in-band FEC (LBRR). See the issue for details
* Added a OPUS_SET_INBAND_FEC(2) option that turns on FEC, but does not force SILK mode (FEC will be disabled in CELT mode)
* Improved tuning and various fixes to DTX
* Added Meson support, improved CMake support In addition to the improvements above, this release includes many minor bug fixes.

==== libopus v1.5.1 ====
Released on March 4th, 2024. This version contains:

* Significant improvement to packet loss robustness using Deep Redundancy (DRED)
* Improved packet loss concealment through Deep PLC
* Low-bitrate speech quality enhancement down to 6 kb/s wideband
* Improved x86 (AVX2) and Arm (Neon) optimizations
* Support for 4th and 5th order ambisonics

==== libopus v1.6 ====
Released on December 15th, 2025. This version contains:

* A new wideband-to-fullband bandwidth extension (BWE) module
* Support for 96 kHz audio with Opus HD
* Significant improvement to Deep Redundancy (DRED)
* A new 24-bit encoder/decoder API
* Fixed-point improvements

=== Other implementations ===

==== Concentus ====

The libopus reference library (fixed-point variant) has successfully been ported to both '''C#''' and '''Java''', as part of a project called '''Concentus'''. The aim of the project is specifically to target cross-platform applications where native C interop is relatively difficult. The code is available on [https://github.com/lostromb/concentus Github] and distributed via standard package managers.

==== Emscripten ports ====

At least one port of reference opus in Javascript has been made using the automated tool [https://developer.mozilla.org/en-US/docs/Mozilla/Projects/Emscripten emscripten]. See [https://blog.rillke.com/opusenc.js/ here], [https://github.com/kazuki/opus.js-sample here] and [https://github.com/audiocogs/opus.js here].

==== ffmpeg ====
FFmpeg has a native [https://ffmpeg.org/ffmpeg-codecs.html#opus "opus"] codec. It is of lower quality than the reference libopus and only does CELT coding. However, it is still good for the ecosystem to have a completely independent implementation.

== Hardware & Software Support ==

Much of this section is based heavily on the Jan 12th 2013 version of the '''Support''' section of the [http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Wikipedia article], which is more likely to be kept updated and to provide links to further information about the supporting platforms.

=== VoIP software ===
* The open source virtual PBX Freeswitch supports Opus transcoding.
* The voice-chat software Mumble supports Opus as its main codec.
* SIP softphones Phoner and PhonerLite support Opus
* The SIP and IAX2 client SFLphone is being fitted with Opus support.
* Integration of Opus into the Skype client is finished, although no version with Opus support has yet been published.
* TrueConf video conferencing solutions support Opus.
* Opus support is planned for Jitsi 2.0, together with VP8 video.
* Empathy may use any format supported in GStreamer, including Opus.
* Line2 has replaced their current codec with Opus. Their iOS app will be the first to be released with the Opus. The Android app will follow later.
* CSipSimple supports Opus, Codec2, G.726 and G.722.1 with an additional plug-in.
* The voice-chat software TeamSpeak 3 supports Opus for voice and music in pre-release server 3.0.7-pre2 and beta client version 3.0.10.
* The proprietary instant messenger service Discord uses Opus audio for all voice calls and video calls, regardless of platform.

=== Web frameworks and browsers ===
* Opus support is mandatory for WebRTC implementations.
* Mozilla supports Opus beginning with version 15 of Firefox and Thunderbird, plus Seamonkey, which uses a shared codebase.
* Depending on the backend in use, Opera supports inline playback of embedded Opus files. Official support for Opus and WebRTC are on the development roadmap.
* Chromium and Google Chrome have audio support as of version 33.
* Apple's Safari browser now supports Opus as of iOS 11 and macOS 10.13 High Sierra.
* Maxthon Cloud Browser

=== Streaming audio ===
* Icecast. (examples: [http://dir.xiph.org/by_format/Opus Stream directory by format Opus], [http://smj.delfa.net/opus_64.m3u 64k]/[http://smj.delfa.net/opus_256.m3u 256k] [http://smj.delfa.net/ Smooth Jazz Opus Stream], [http://www.absoluteradio.co.uk/listen/labs.html Absolute Radio Opus Trial] 7 stations at 24,64,96 kbps, [http://icecast.ofdoom.com:8000/burst-opus.ogg Icecast Of Doom 96k]
* Krad Radio
* Liquidsoap

=== Operating systems and desktop multimedia frameworks ===
* In Debian GNU/Linux the Opus development tools and supporting libraries can be installed from the preconfigured repositories in the next stable version ("wheezy") that is expected to be released in early 2013.
* For Microsoft Windows, there are DirectShow filters supporting Opus, including DC-Bass Source Mod and the LAV Filters.
* In GStreamer the integration of Opus support is complete.
* FFmpeg supports decoding and encoding Opus via the external library libopus.
* Android 5.0 and above supports Opus natively if encapsulated in the Ogg container, but .opus filename extension is not recognized by Android, so the use of double filename extension .opus.ogg is recommended as a workaround to allow apps to recognize files as playable audio.

=== Hardware support ===
* Support in [[Rockbox]] is available. This means hardware support for a series of portable media players (including some products from the iPod series by Apple and Sansa, iriver and Archos devices) and with "Rockbox as an Application" (RaaA) also on Android devices.

=== Player software ===

* Windows/Mac/Linux (Cross-Platform)
*# [[VLC]] (media player supports Opus as of version 2.0.4
*# [[Amarok]] 2.8 has transcoding support for Opus codec if ffmpeg is compiled with support for the libopus library & support for playback of Opus encoded files if Amarok is compiled against TagLib (newer than V1.8)
*# Clementine has Opus support
*# Audacious player
*# [[MPD]] as of version 0.18 if compiled against libopus (supports both encoding for http streams and decoding)
* Windows Exclusive
*# AIMP supports Opus natively as of version 3.20 build 1125 beta 1
*# [[foobar2000]] supports Opus natively as of v1.1.14 beta 1
*# Mpxplay supports Opus (using a decoder DLL) as of v1.60 alpha 2
*# [[Winamp]] supports Opus using a [http://forums.winamp.com/showthread.php?p=2925154#post2925154 3rd party plug-in]
*# MPC-HC
*# Resonic Player/Pro supports Opus natively as of version 0.2.2
* iOS/Android (Cross-Platform)
*# Capriccio [https://itunes.apple.com/us/app/capriccio-free-ultimate-music/id434829018?mt=8 iOS]/[https://play.google.com/store/apps/details?id=me.ideariboso.capriccio Android]
*# foobar2000 [https://itunes.apple.com/us/app/foobar2000/id1072807669?mt=8 iOS]/[https://play.google.com/store/apps/details?id=com.foobar2000.foobar2000&hl=en Android]
* Android Exclusive
*# [https://play.google.com/store/apps/details?id=in.krosbits.musicolet Musicolet Music Player]
*# [http://gonemadmusicplayer.blogspot.com/ GoneMAD Music Player]
*# [http://neutronmp.com/ Neutron Music Player]
*# [http://www.videolan.org/vlc/download-android.html VLC Media Player for Android]
*# [https://play.google.com/store/apps/details?id=ru.recoilme.freeamp FreeMP]
*# [https://play.google.com/store/apps/details?id=net.mderezynski.youki3 Youki]
*# [https://play.google.com/store/apps/details?id=com.aimp.player AIMP for Android]
*# [https://play.google.com/store/apps/details?id=com.acmeandroid.listen Listen Audiobook Player]
*# [https://play.google.com/store/apps/details?id=com.mxtech.videoplayer.ad MX Player]
*# [https://play.google.com/store/apps/details?id=org.tomahawk.tomahawk_android Tomahawk Player Beta]
*# [https://play.google.com/store/apps/details?id=com.maxmpz.audioplayer&hl=en Poweramp Music Player]

=== Other software ===
* CDBurnerXP
* MediaCoder
* Report-IT
* [[MP3tag|MP3tag]]
* [https://moisescardona.me/opus-gui/ Opus GUI]
* [http://www.xdlab.ru/en/ TagScanner]
* [http://www.xmedia-recode.de/ XMedia Recode]
* [[loudgain]]

== References & Notes ==

*{{note|homepage|a}}[http://opus-codec.org/ opus-codec.org homepage]
*{{note|FAQ|b}}[http://wiki.xiph.org/OpusFAQ Opus FAQ]
*{{note|RFC|c}}[http://tools.ietf.org/html/rfc6716 IETF RFC 6716]
<references/>
[[Category:Codecs]]
[[Category:Lossy]]
[[Category:Encoder/Decoder]]

Foobar2000:Free Encoder Pack

2026-01-15T23:03:44Z

Logan589XP: Updated encoder version numbers and release dates to reflect the newest Free Encoder Pack release.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2026-01-15
| stable_release_date = 2026-01-15
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.85
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.85
|
|-
| [[FLAC]]
| flac
| 1.5.0
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100 (q4 patched)
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.6.1
|-
| [[WavPack]]
| wavpack
| 5.8.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Foobar2000:Free Encoder Pack

2025-06-26T19:29:12Z

Logan589XP: Updated encoder version numbers and release dates to reflect the newest Free Encoder Pack release.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2025-05-07
| stable_release_date = 2025-05-07
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.85
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.85
|
|-
| [[FLAC]]
| flac
| 1.5.0
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.5.2
|-
| [[WavPack]]
| wavpack
| 5.8.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Apple AAC

2025-06-23T22:52:39Z

Logan589XP: Updated version number to 2.85.

{{aac-encoders}}
'''Apple AAC''' is an [[AAC]] codec developed by Apple Inc. and included in Apple's audio framework, ''Core Audio''. ''Core Audio'' is used in [[iTunes]], Safari web browser, and other Apple products. Apple's AAC encoder is known to be one of the highest quality medium-bitrate [[CBR]] and [[VBR]] LC [[AAC]] encoders. The encoder started as part of the ''QuickTime'' media framework, but Apple has transitioned away from ''QuickTime'', to ''AV Foundation'' and ''AudioToolbox''.

The encoder does consistently well in [[Hydrogenaudio Listening Tests|Hydrogenaudio listening tests]].
__TOC__
== Bitrate modes ==
Four rate settings are present in Apple AAC:<ref>https://hydrogenaud.io/index.php/topic,118203.0.html Testing by Pr0m3th3u5 in 2019; comment by JAZ</ref>
; Constant bitrate (CBR)
: Similar to MP3 CBR in how it handles the bit reservior, but can still swing a little from the target
; Average bit rate (ABR)
: Self-explanatory
; Constrained VBR
: Like ABR, but seems to be more open to using more bits
; Variable bitrate (VBR)
; "True" VBR (TVBR, unofficial name)
: Takes a quality from 0 to 127 (kAudioCodecPropertySoundQualityForVBR). Seems to perform similarly to cVBR once the below table is used for bitrate conversion.

{|class=wikitable
|+TVBR bitrate mapping (AAC)
|-
! -ue / -Q !! ffmpeg q !! bitrate (bps)
|-
| 45 || 9.11 || 96
|-
| 64 || 7 || 128
|-
| 73 || 6 || 144
|-
| 82 || 5 || 160
|-
| 91 || 4 || 192
|-
| 109 || 2 || 256
|-
| 127 || 0 || 320
|}

The above is approximated by a piece-wise function, although the neatness is likely more indicative of how HA users pick values for tests:

Bitrate(q) = (40 * q - 1184) / 9 82 <= q < 128
(16 * q + 128) / 9 64 <= q < 82

TVBR is not available for HE-AAC.

== Format support ==
Apple's CoreAudioBaseTypes lists the following AAC-derived codecs on macOS 11.3:<ref name=CABT>https://github.com/phracker/MacOSX-SDKs/blob/041600eda65c6a668f66cb7d56b7d1da3e8bcc93/MacOSX11.3.sdk/System/Library/Frameworks/CoreAudioTypes.framework/Versions/A/Headers/CoreAudioBaseTypes.h#LL391C2-L391C</ref>
* AAC-LC (MPEG-4)
* AAC-LD
* AAC-ELD
* AAC-ELD + SBR
* HE-AAC v1 (AAC + SBR)
* HE-AAC v2 (v1 + PS)
* USAC
* MPEG-4 Spatial

There should also be support for a number of other format, at least in decoding: Apple does not ship a standalone AAC codec, but a whole audio-processing kit.

Actual support depends on library version and whether the program used to call the library knows how to use these formats.
* afconvert is clumsy but exposes almost everything.
* QAAC only does LC and HEv1. It can use some of the audiotoolkit filters while it works.
* FFmpeg only knows about using Apple AAC with LC, LD, ELD, HE, and HEv2.

== afconvert ==
Apple's macOS includes a command-line utility for transcoding audio files that employs this encoder when encoding to AAC.

Usage:
afconvert [options] <infile> <outfile>

Options:
;-f --file <string>:File format. Several; use 'm4af' for '.m4a' file or 'adts' for raw '.aac' file.
;-d --data <string>:Data format. Several; use 'aac' for AAC-LC. See CoreAudioBaseTypes.h link in footnote.<ref name=CABT/>
;-s --strategy <0,1,2,3>:Bitrate allocation strategy. 0 for CBR, 1 for ABR, 2 for VBR_constrained, 3 for VBR
;-b --bitrate <n>:Bitrate in bits per second. Does nothing when used with strategy 3.
;-ue vbrq <0-127>:Output quality for VBR mode. Only applicable when using strategy 3.
;-q --quality <0-127>:Speed/quality trade-off. Internally rounded to a value of either 32, 64, or 96.

afconvert has way more options than those listed above, but not all are useful for AAC. Run <code>afconvert --help</code> to see them all, or see [https://ss64.com/osx/afconvert.html SS64] for an unofficial list.

== QAAC ==
{{Software Infobox
| name = qaac
| logo =
| screenshot =
| caption =
| maintainer = nu774
| stable_release = 2.85
| preview_release =
| operating_system = Windows
| use = Encoder
| license = Various
| website = [https://github.com/nu774/qaac github/qaac]
}}

QAAC is an open-source wrapper for ''Core Audio's'' AAC and [[ALAC]] encoders that allow them to be used directly in applications that don't otherwise use ''Core Audio''. It exists only for Windows.

* [https://github.com/nu774/qaac/wiki QAAC documentation]

Linux users [https://www.andrews-corner.org/qaac.html should be able to use wine]. The most fragile part apparently is [https://github.com/enzo1982/freac/issues/168 getting AppleApplicationSupport to install].

 

== ffmpeg ==
On macOS, ffmpeg 4.4 and above can use the <code>aac_at</code> codec to use Apple AAC via the new AudioToolbox.framework. In addition to AAC-LC, HE-AAC (v1/v2), AAC-LD, and AAC-ELD are available, the same selection of profiles as FDK AAC. FFmpeg uses a 0-14 quality range to be converted to the 0-127 scale, using the expression <code>q = 127 - q * 9;</code>.<ref>https://github.com/FFmpeg/FFmpeg/blob/f009f849617ab0c2015d2a748892d55b37d378c0/libavcodec/audiotoolboxenc.c#L328</ref>

On Windows, ffmpeg's build process can be tricked into linking to an CoreAudioToolbox.dll commonly distributed alongside iTunes, using [https://github.com/dantmnf/AudioToolboxWrapper AudioToolboxWrapper].

==External links==
<references />
*[http://www.apple.com/quicktime Apple's Quicktime Website]

Foobar2000:Foobar2000

2025-02-18T20:39:35Z

Logan589XP: Updated stable release version number and release date.

{{Software Infobox|
|name = foobar2000
|logo = [[Image:foobar2000 Logo.png|48px]]
|screenshot = [[Image:Foobar2000-1.0-default-ui.png|250px]]
|caption = Screenshot of foobar2000 v1.0 using the default user interface
|developer = Peter Pawlowski
|released = {{start date and age|2002|12|20}}<ref>{{ha|https://hydrogenaud.io/index.php/topic,4997.0.html|Official Foobar2000 site & Foobar2000 0.3 & SDK!}}</ref>
|stable_release = 2.24.2<ref>{{foobar2000|https://www.foobar2000.org/download|Download foobar2000 for Windows}}</ref>
|stable_release_date = 2025-02-18
|preview_release =
|preview_release_date =
|operating_system = Windows
|use = Media Player
|license = Proprietary, BSD
|website = [https://www.foobar2000.org/ foobar2000.org]
}}
'''foobar2000''' is an advanced freeware audio player for the Windows platform. Some of the basic features include full Unicode support, [[ReplayGain]] support and native support for several popular audio formats.

== Platforms ==

foobar2000 has been written specifically for the Windows platform, and there are no plans to port it to any others. However, although not officially supported, it is known to run on [https://www.hydrogenaudio.org/forums/index.php?showtopic=54933 Linux] and [https://www.hydrogenaudio.org/forums/index.php?showtopic=77261 Mac OS X] through Wine and WineBottler, respectively. Also there is an [https://aur.archlinux.org/packages/foobar2000 Arch Linux native package] maintained by a community member.

== Features ==
* Powerful open component architecture allowing third-party developers to extend functionality of the player, including the ability to fully replace the user interface.
* Full Unicode support: File names, user interface, tagging, etc.
* [[ReplayGain]] support: Both playback and writing ReplayGain information to file tags.
* [[Gapless playback]].
* Advanced [[tagging]] capabilities - through built-in [[foobar2000:Properties|Properties dialog]] and various optional tagging-related components.
* Built-in [[foobar2000:Preferences:Media Library|Media Library]] functionality.
** Intuitive [[foobar2000:Query syntax|query syntax]] for searching the Media Library.
** Autoplaylist support: Generate dynamically updating playlists based on queries.
* [[foobar2000:Preferences:General:Keyboard Shortcuts|Customizable keyboard shortcuts]].
* Support for [[transcoding]] all supported audio formats using the [[Foobar2000:Converter|Converter component]] (requires external command-line encoder executables for different output formats).
* [[Secure_ripping|Secure]] [[foobar2000:Ripping CDs|CD ripping]].
* Streaming support.
* Efficient handling of large playlists.
* [[foobar2000:Components/Default_user_interface_%28foo_ui_std%29|User interface]] with simple configuration to create even complex layouts quickly and easily.
* Highly customizable display of track information using [[foobar2000:Titleformat_Introduction|title formatting scripts]].

==Supported audio formats==
Natively supported ("out-of-the-box"):
* [[MP1]], [[MP2]], [[MP3]], [[MP4]], [[Musepack]], [[AAC]], [[Ogg Vorbis]], [[FLAC]] / Ogg FLAC, [[Speex]], [[WavPack]], [[WAV]], [[AIFF]], [[AU|AU/SND]], [[CDDA]], [[WMA]], [[Matroska]], [[ALAC]], [[MMS]], [[RTSP]], [[Opus]], [[Monkey's Audio]], [[AC3]], [[DTS]], [[TAK]].
Supported through optional components:
* [[TTA]], [[MOD]], [[SPC]], [[Shorten]], [[OptimFROG]], [[PSF]], [[NSF]], [[XID]], [[XA]], [[AMR]], ''etc.''

In addition, foobar2000 can also play audio files within ZIP and RAR archives (typically compressed) directly, ''i.e.'' without requiring the user to extract the files first. More archive formats are supported ''via'' additional components: [https://www.foobar2000.org/components/view/foo_unpack_lha LHA/LZH], [https://www.foobar2000.org/components/view/foo_unpack_7z 7-Zip]

== Using foobar2000 ==

=== Important pages ===
* {{foobar2000|https://www.foobar2000.org/FAQ|foobar2000 FAQ}}
* [[foobar2000:components|foobar2000 components]]
* [[foobar2000:Legacy components|foobar2000 legacy components]]
* [[foobar2000:Encouraged Tag Standards|foobar2000 encouraged tag standards]]
* [[foobar2000:FAQ|foobar2000 FAQ (unofficial)]]
* [[:Category:Foobar2000_Guides|foobar2000 guides (category)]]
* [[:Category:Foobar2000_Preferences|foobar2000 preferences (category)]]

=== Specific guides ===

'''Preferences'''
* [[foobar2000:Preferences|Preferences dialog]]

'''Metadata'''
* [[foobar2000:Properties|Tag editing: the Properties dialog]]
* [[foobar2000:Query syntax|Query Syntax]]: details of Syntax for querying metadata.

'''Title formatting'''
* [[foobar2000:Title Formatting Introduction|Introduction to titleformat scripts]]
* [[foobar2000:Title Formatting Reference|Titleformat Reference]]: reference guide to all fields and functions
* [[foobar2000:Titleformat Examples|Titleformat Examples]]: user-submitted code for various purposes; submit your own!

'''Others'''
* [[foobar2000:File operations|File operations dialog]]: move, copy, rename, and delete files from within foobar2000
* [[foobar2000:Commandline Guide|Commandline usage]]

=== External Guides ===
* [https://www.audiohq.de/viewforum.php?id=27 German-language guides]
* [https://foobar2000.xrea.jp/ fb2k Wiki Page] for Japanese users.
* [https://winamp2foobar.blogspot.com Winamp To Foobar Guide] with information relevant for general users also.

=== Technical information ===
* [[foobar2000:ID3 Tag Mapping|ID3 Tag Mapping]]
* [[foobar2000:Metadata Compatibility|Metadata Compatibility]]: information about compatibility with metadata written by other applications

== Important Links ==
=== Official site ===
* [https://www.foobar2000.org foobar2000.org: Homepage]
* [https://www.foobar2000.org/download foobar2000.org: Download]
* [https://www.foobar2000.org/components foobar2000.org: Components]

=== Community ===
* Official IRC Channel: #foobar2000 on [https://libera.chat/ Libera Chat]
* {{ha|https://hydrogenaud.io/index.php/board,28.0.html|foobar2000 Forums}}
* [https://foobar-users.de/ German Support Forum]
* [http://www.fforum.ru/viewforum.php?f=59 Russian-language forum]
* [https://www.foobar2000.ru/forum/ Another Russian-language forum]

=== Appearance ===
* {{ha|https://hydrogenaud.io/index.php/topic,61333.0.html|Default UI .fth thread}} (fast way to clone another's DUI configuration)
* [[foobar2000:Preferences:Columns UI/Appearance|Columns UI appearance customization guides]]
* {{ha|https://hydrogenaud.io/index.php/topic,31027.0.html|Columns UI configurations}}

== References ==
<references/>

[[Category:foobar2000]]
[[Category:Media Players|foobar2000]]
[[Category:CD Rippers]]
[[Category:Software]]
[[Category:Tag editors]]

Foobar2000:Foobar2000

2025-02-17T09:26:47Z

Logan589XP: Updated stable release version number and release date.

{{Software Infobox|
|name = foobar2000
|logo = [[Image:foobar2000 Logo.png|48px]]
|screenshot = [[Image:Foobar2000-1.0-default-ui.png|250px]]
|caption = Screenshot of foobar2000 v1.0 using the default user interface
|developer = Peter Pawlowski
|released = {{start date and age|2002|12|20}}<ref>{{ha|https://hydrogenaud.io/index.php/topic,4997.0.html|Official Foobar2000 site & Foobar2000 0.3 & SDK!}}</ref>
|stable_release = 2.24.1<ref>{{foobar2000|https://www.foobar2000.org/download|Download foobar2000 for Windows}}</ref>
|stable_release_date = 2024-12-16
|preview_release =
|preview_release_date =
|operating_system = Windows
|use = Media Player
|license = Proprietary, BSD
|website = [https://www.foobar2000.org/ foobar2000.org]
}}
'''foobar2000''' is an advanced freeware audio player for the Windows platform. Some of the basic features include full Unicode support, [[ReplayGain]] support and native support for several popular audio formats.

== Platforms ==

foobar2000 has been written specifically for the Windows platform, and there are no plans to port it to any others. However, although not officially supported, it is known to run on [https://www.hydrogenaudio.org/forums/index.php?showtopic=54933 Linux] and [https://www.hydrogenaudio.org/forums/index.php?showtopic=77261 Mac OS X] through Wine and WineBottler, respectively. Also there is an [https://aur.archlinux.org/packages/foobar2000 Arch Linux native package] maintained by a community member.

== Features ==
* Powerful open component architecture allowing third-party developers to extend functionality of the player, including the ability to fully replace the user interface.
* Full Unicode support: File names, user interface, tagging, etc.
* [[ReplayGain]] support: Both playback and writing ReplayGain information to file tags.
* [[Gapless playback]].
* Advanced [[tagging]] capabilities - through built-in [[foobar2000:Properties|Properties dialog]] and various optional tagging-related components.
* Built-in [[foobar2000:Preferences:Media Library|Media Library]] functionality.
** Intuitive [[foobar2000:Query syntax|query syntax]] for searching the Media Library.
** Autoplaylist support: Generate dynamically updating playlists based on queries.
* [[foobar2000:Preferences:General:Keyboard Shortcuts|Customizable keyboard shortcuts]].
* Support for [[transcoding]] all supported audio formats using the [[Foobar2000:Converter|Converter component]] (requires external command-line encoder executables for different output formats).
* [[Secure_ripping|Secure]] [[foobar2000:Ripping CDs|CD ripping]].
* Streaming support.
* Efficient handling of large playlists.
* [[foobar2000:Components/Default_user_interface_%28foo_ui_std%29|User interface]] with simple configuration to create even complex layouts quickly and easily.
* Highly customizable display of track information using [[foobar2000:Titleformat_Introduction|title formatting scripts]].

==Supported audio formats==
Natively supported ("out-of-the-box"):
* [[MP1]], [[MP2]], [[MP3]], [[MP4]], [[Musepack]], [[AAC]], [[Ogg Vorbis]], [[FLAC]] / Ogg FLAC, [[Speex]], [[WavPack]], [[WAV]], [[AIFF]], [[AU|AU/SND]], [[CDDA]], [[WMA]], [[Matroska]], [[ALAC]], [[MMS]], [[RTSP]], [[Opus]], [[Monkey's Audio]], [[AC3]], [[DTS]], [[TAK]].
Supported through optional components:
* [[TTA]], [[MOD]], [[SPC]], [[Shorten]], [[OptimFROG]], [[PSF]], [[NSF]], [[XID]], [[XA]], [[AMR]], ''etc.''

In addition, foobar2000 can also play audio files within ZIP and RAR archives (typically compressed) directly, ''i.e.'' without requiring the user to extract the files first. More archive formats are supported ''via'' additional components: [https://www.foobar2000.org/components/view/foo_unpack_lha LHA/LZH], [https://www.foobar2000.org/components/view/foo_unpack_7z 7-Zip]

== Using foobar2000 ==

=== Important pages ===
* {{foobar2000|https://www.foobar2000.org/FAQ|foobar2000 FAQ}}
* [[foobar2000:components|foobar2000 components]]
* [[foobar2000:Legacy components|foobar2000 legacy components]]
* [[foobar2000:Encouraged Tag Standards|foobar2000 encouraged tag standards]]
* [[foobar2000:FAQ|foobar2000 FAQ (unofficial)]]
* [[:Category:Foobar2000_Guides|foobar2000 guides (category)]]
* [[:Category:Foobar2000_Preferences|foobar2000 preferences (category)]]

=== Specific guides ===

'''Preferences'''
* [[foobar2000:Preferences|Preferences dialog]]

'''Metadata'''
* [[foobar2000:Properties|Tag editing: the Properties dialog]]
* [[foobar2000:Query syntax|Query Syntax]]: details of Syntax for querying metadata.

'''Title formatting'''
* [[foobar2000:Title Formatting Introduction|Introduction to titleformat scripts]]
* [[foobar2000:Title Formatting Reference|Titleformat Reference]]: reference guide to all fields and functions
* [[foobar2000:Titleformat Examples|Titleformat Examples]]: user-submitted code for various purposes; submit your own!

'''Others'''
* [[foobar2000:File operations|File operations dialog]]: move, copy, rename, and delete files from within foobar2000
* [[foobar2000:Commandline Guide|Commandline usage]]

=== External Guides ===
* [https://www.audiohq.de/viewforum.php?id=27 German-language guides]
* [https://foobar2000.xrea.jp/ fb2k Wiki Page] for Japanese users.
* [https://winamp2foobar.blogspot.com Winamp To Foobar Guide] with information relevant for general users also.

=== Technical information ===
* [[foobar2000:ID3 Tag Mapping|ID3 Tag Mapping]]
* [[foobar2000:Metadata Compatibility|Metadata Compatibility]]: information about compatibility with metadata written by other applications

== Important Links ==
=== Official site ===
* [https://www.foobar2000.org foobar2000.org: Homepage]
* [https://www.foobar2000.org/download foobar2000.org: Download]
* [https://www.foobar2000.org/components foobar2000.org: Components]

=== Community ===
* Official IRC Channel: #foobar2000 on [https://libera.chat/ Libera Chat]
* {{ha|https://hydrogenaud.io/index.php/board,28.0.html|foobar2000 Forums}}
* [https://foobar-users.de/ German Support Forum]
* [http://www.fforum.ru/viewforum.php?f=59 Russian-language forum]
* [https://www.foobar2000.ru/forum/ Another Russian-language forum]

=== Appearance ===
* {{ha|https://hydrogenaud.io/index.php/topic,61333.0.html|Default UI .fth thread}} (fast way to clone another's DUI configuration)
* [[foobar2000:Preferences:Columns UI/Appearance|Columns UI appearance customization guides]]
* {{ha|https://hydrogenaud.io/index.php/topic,31027.0.html|Columns UI configurations}}

== References ==
<references/>

[[Category:foobar2000]]
[[Category:Media Players|foobar2000]]
[[Category:CD Rippers]]
[[Category:Software]]
[[Category:Tag editors]]

Foobar2000:Free Encoder Pack

2025-02-17T09:11:39Z

Logan589XP: Added missing ending 0 in FLAC version number.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2025-02-11
| stable_release_date = 2025-02-11
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.83
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.83
|
|-
| [[FLAC]]
| flac
| 1.5.0
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.5.2
|-
| [[WavPack]]
| wavpack
| 5.8.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Free Lossless Audio Codec

2025-02-17T09:04:44Z

Logan589XP: Updated to version 1.5.0

{{Codec Infobox
| name = FLAC
| logo = [[Image:FLAC logo.gif]]
| type = lossless
| purpose = Popular open source patent free lossless compression scheme.
| maintainer = Josh Coalson, Xiph Community
| released = {{start date and age|2001}}
| recommended_encoder = FLAC encoder
| recommended_text = FLAC v1.5.0 (11 Feb 2025)
| website = http://xiph.org/flac/
}}
'''Free Lossless Audio Codec''' ('''FLAC''') is a codec for lossless audio compression.
Grossly oversimplified, FLAC is similar to [[MP3]], but [[lossless]], meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player (or your car or home stereo, if supported) just like you would a lossy file.

FLAC is natively supported in Android since version 3.1, in Windows 10, and in macOS High Sierra 10.13.

== General aspects of the format ==
FLAC is freely available and supported on most operating systems, including Windows, UNIX (Linux, *BSD, Solaris, OS X, IRIX), BeOS, OS/2, and Amiga. There are build systems for autotools, MSVC, Watcom C, and Project Builder.

The FLAC project consists of:
* the stream format
* reference encoders and decoders in library form
* flac, a command-line program to encode and decode FLAC files
* metaflac, a command-line metadata editor for FLAC files
* input plugins for various music players

When it's said that FLAC is ''free'', it means more than just that it is available at no cost. It means that the specification of the format is fully open to the public to be used for any purpose, although the FLAC project reserves the right to set the FLAC specification and certify compliance. It also means that neither the FLAC format nor any of the implemented encoding/decoding methods are covered by any known patent. And it means that all the source code is available under [http://xiph.org/flac/license.html open-source licenses]. It is the first truly open and free lossless audio format.

== Features ==
* '''Lossless:''' The encoding of audio (PCM) data incurs no loss of information, and the decoded audio is bit-for-bit identical to what went into the encoder. Each frame contains a 16-bit CRC of the frame data for detecting transmission errors. The integrity of the audio data is further insured by storing an MD5 signature of the original unencoded audio data in the file header, which can be compared against later during decoding or testing.
* '''Fast:''' FLAC is asymmetric in favor of decode speed. Decoding requires only integer arithmetic, and is much less compute-intensive than for most perceptual codecs. Real-time decode performance is easily achievable on even modest hardware.
* '''Hardware support:''' Because of FLAC's free reference implementation, low decoding complexity and popularity, FLAC has relatively widespread hardware support compared to other lossless formats.
* '''Streamable:''' Each FLAC frame contains enough data to decode that frame. FLAC does not even rely on previous or following frames. FLAC uses sync codes and CRCs (similar to MPEG and other formats), which, along with framing, allow decoders to pick up in the middle of a stream with a minimum of delay.
* '''Seekable:''' FLAC supports fast sample-accurate seeking. Not only is this useful for playback, it makes FLAC files suitable for use in editing applications.
* '''Flexible metadata:''' New metadata blocks can be defined and implemented in future versions of FLAC without breaking older streams or decoders. Currently there are metadata types for tags, cue sheets, and seek tables. Applications can write their own APPLICATION metadata once they register an ID.
* '''Suitable for archiving:''' FLAC is an open format, and there is no generation loss if you need to convert your data to another format in the future. In addition to the frame CRCs and MD5 signature, FLAC has a verify option that decodes the encoded stream in parallel with the encoding process and compares the result to the original, aborting with an error if there is a mismatch.
* '''Convenient CD archiving:''' FLAC has a ''cue sheet'' metadata block for storing a CD table of contents and all track and index points. For instance, you can rip a CD to a single file, then import the CD's extracted cue sheet while encoding to yield a single file representation of the entire CD. If your original CD is damaged, the cue sheet can be exported later in order to burn an exact copy.
* '''Error resistant:''' Because of FLAC's framing, stream errors limit the damage to the frame in which the error occurred, typically a small fraction of a second worth of data. Contrast this with some other lossless codecs, in which a single error destroys the remainder of the stream.

== Pros ==
* Portable to many systems
* Open source and freely licensed
* Hardware support (PhatBox, Kenwood MusicKeg, Rio Karma, etc. See below)
* Streaming support
* Extremely fast decoding
* Supports multichannel and high resolution streams
* Supports [[ReplayGain]]
* Supports cue-sheet (with some limitations)
* Gaining wide use as successor to [[Shorten]]

== Cons ==
* Compresses less efficiently than other popular modern compressors ([[Monkey's Audio]], [[OptimFROG]])
* Higher compression modes slow, for little gain over the default setting.
* Only integer bit depth (up to 32-bit int) format are supported, or in other words, no floating-point support.

== Hardware and software that support FLAC ==
For a more comprehensive list see the [http://xiph.org/flac/links.html FLAC links page].

=== Hardware ===
==== Car stereo ====
* Kenwood [http://www.kenwood.com/cs/ce/audiofile/index.php?model=KMM KMM series]
* Pioneer [http://www.pioneer.eu/eur/products/25/121/61/overview.html Car Stereo] (search FLAC)
* JVC [http://mobile.jvc.com/product.jsp?pathId=139 KD-X "Digital Media Receivers" series] (almost all)
* Soundstream [http://soundstream.com/store/car-video/source-units.html Source Units]
* Tesla Model S
* Citroën DS5
* Volvo Sensus (SPA Platform)

==== Home stereo ====
* Olive's [http://www.olive.us/ Symphony] wireless digital music center
* [http://www.numark.com/ Numark]'s DJ equipment (HDX and CDX turntables, HDMIX mixer)
* [http://www.sonos.com/ Sonos Digital Music System]
* Slim Devices' [http://www.slimdevices.com/pi_squeezebox.html Squeezebox] networked audio players

==== Portable ====
* [[Apple iPod]] with [[Rockbox]] firmware
* [[iAudio M3]], M5 and X5
* [[iRiver]] iHP-120/iHP-140 with [[Rockbox]] firmware
* [[Iwod G10]]
* [[Rio Karma]]
* [http://en.wikipedia.org/wiki/SanDisk_Sansa SanDisk Sansa]
* TrekStor's [http://www.trekstor.de/en/products/detail_mp3.php?pid=66 Vibez]
* Devices running Android 3.1+
* [[Pono|Pono Player]]
* [http://en.wikipedia.org/wiki/FiiO_X_Series FiiO X Series]

== Software ==

==== Players Software ====

*Windows
*# [[foobar2000]]
*# [[MediaMonkey]]
*# [[MusicBee]]
*# [[Winamp]]
*# [https://www.aimp.ru/ AIMP]
*# [[VLC]]
*# [http://www.un4seen.com/ XMPlay]
*# [http://mplayerwin.sourceforge.net/ MPlayer] Console player

*Mac
*# [[foobar2000]]
*# [http://cogx.org/ Cog]
*# [[VLC]]
*# [http://coppertino.com/ VOX]

*Linux
*# [http://www.clementine-player.org/ Clementine]
*# [http://www.mplayerhq.hu/ MPlayer]
*# [http://www.mythtv.org/ MythTV]
*# [[VLC]]
*# [[XMMS]]

*Android
*# [https://powerampapp.com/ Poweramp]
*# [https://www.aimp.ru/ AIMP]
*# [[foobar2000]]
*# [[VLC]]

===Frontends (Windows)===
* [[foobar2000]] Music Converter / Audio Player
* [https://moisescardona.me/flac-gui/ FLAC GUI] (By Moisés Cardona)
* [https://github.com/hat3k/FLAC-Frontend-H FLAC frontend] ({{ha user|u=29445|name=hat3k}} fork) {{ha|https://hydrogenaud.io/index.php/topic,127093.0.html}}
* <s> FLAC frontend (ktf) </s>
* <s> Windows Frontend (by Speek) </s>

===Frontends (Mac)===
* [http://www.sbooth.org/Max/ Max]

===Converters===
*for a more comprehensive list [http://wiki.hydrogenaud.io/index.php?title=Download_page#Transcoders here]
*# [http://www.dbpoweramp.com/ dBpowerAMP] Music Converter / Audio Player / CD Writer
*# [[ MediaMonkey]] Music Manager / Audio Player / CD Writer
*# [[fre:ac]] Music Converter / CD Writer/Ripper
*# [[foobar2000]] Music Converter / Audio Player

===Editors===
*List of popular audio editors
*# [[Audacity]]
*# [[Adobe Audition]]
*# [http://www.goldwave.com/ GoldWave]

===CD writers/rippers===
*for a comprehensive list of CD/DVD [http://wiki.hydrogenaud.io/index.php?title=Download_page#CD.2FDVD_Writers Writers] & [http://wiki.hydrogenaud.io/index.php?title=Download_page#CD_Rippers rippers]
*# [http://www.nero.com/eng/ Nero]
*# [http://cdburnerxp.se/ CDBurnerXP] CD writer
*# [[MediaMonkey]] - CD ripper/writer
*# [[Exact Audio Copy]] CD Ripper
*# [http://www.cdwave.com/ CD Wave]
*# [http://arson.sourceforge.net/ Arson]
*#  Burrrn  Audio CD burner
*#  CDex CD ripper

===Taggers===
*for a comprehensive list of tag editors [[http://wiki.hydrogenaud.io/index.php?title=Download_page#Tagging_Utilities here]]
*# [[Mp3tag]] - Universal Tag Editor
*# [http://www.xdlab.ru/en/ TagScanner] - "The Ultimate Tag Editor"
*# [[foobar2000]]
*# [[MediaMonkey]] - Tagger / Music manager (Including multiple and linked album art support)
*# [http://www.jtclipper.eu/thegodfather/ The GodFather] - Tagger / Music manager
*# [http://sbooth.org/Tag/ Tag] - for Mac OS X 10.4 (Tiger)
*# <s>[http://www.synthetic-soul.co.uk/tag/ Case's Tag] - Command line tagger </s>
*# [https://xiph.org/flac/documentation_tools_metaflac.html metaflac] - for general metadata (including Vorbis comments) maintenance

==== Other tools ====
* [http://www.bunkus.org/videotools/mkvtoolnix/ mkvtoolnix] - tool to multiplex FLAC streams inside the Matroska container
* [https://xiph.org/flac/documentation_tools_metaflac.html metaflac] - for general metadata (including Vorbis comments) maintenance, also to calculate [[ReplayGain]] values for FLAC files lacking such

== Frequently asked questions ==
''Question:'' Does the compression level affect decompression speed?

''Short Answer'': No.

''Long Answer'': In truth, the compression level does affect the decompression speed, but the difference between the various compress levels can barely be measured and is too small to be noticed, even on low-end machines.

''Question:'' What is the best compression level for encoding my music?

''Short Answer'': The default setting, 5.

''Long Answer'': Encoding at the default setting will give the best balance between compression and encoding speed. Encoding at 8 can more than quadruple the encoding time, while having an insignificant effect on compression.

== See also ==
* [[Lossless]]
* [[Lossless comparison]]
* [[FLAC decoder testbench]]

== External links ==
* [http://xiph.org/flac/ FLAC homepage]
* {{ha|http://hydrogenaud.io/index.php/board,67.0.html|FLAC discussion board}}
* {{ha|https://hydrogenaud.io/index.php/topic,123025.0.html|FLAC v1.4.x Performance Tests}}
* {{ha|https://hydrogenaud.io/index.php/topic,124356.0.html|FLAC 1.4.3 topic}}
* {{ha|https://hydrogenaud.io/index.php/topic,123234.0.html|FLAC 1.4.2 topic}}
* {{ha|https://hydrogenaud.io/index.php/topic,122949.0.html|FLAC 1.4.0 topic}}
* {{ha|https://hydrogenaud.io/index.php/topic,118008.msg974100/topicseen.html#new|FLAC 1.3.3 topic}}
* {{ha|http://hydrogenaud.io/index.php/topic,107611.0.html|FLAC 1.3.1 topic}}
* {{ha|http://hydrogenaud.io/index.php/topic,107913.0.html|FLAC 1.3.1 non-SSE2 build for older CPUs}}
* {{ha|http://hydrogenaud.io/index.php/topic,107990.0.html|Lossless codec comparison (Jan ’15)}} by [http://hydrogenaud.io/index.php?action=profile;u=68263 ktf] graphs the influence of the chosen encoding level on the encoding and decoding performance of FLAC 1.3.1 and various other lossless codecs. Omion's older test, "[http://web.archive.org/web/20091108104748/http://people.ucsc.edu/~rswilson/flactest File Size vs. Decoding Speed]", covers the influence of the chosen encoding level on the decoding speed of FLAC 1.2.1.

[[Category:Codecs]]
[[Category:Lossless]]
[[Category:Encoder/Decoder]]

Foobar2000:Free Encoder Pack

2025-02-17T09:00:30Z

Logan589XP: Updated encoder version numbers and release dates to reflect the newest Free Encoder Pack release.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2025-02-11
| stable_release_date = 2025-02-11
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.83
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.83
|
|-
| [[FLAC]]
| flac
| 1.5
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.5.2
|-
| [[WavPack]]
| wavpack
| 5.8.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Apple AAC

2025-01-14T20:37:54Z

Logan589XP: Updated version number to 2.83.

{{aac-encoders}}
'''Apple AAC''' is an [[AAC]] codec developed by Apple Inc. and included in Apple's audio framework, ''Core Audio''. ''Core Audio'' is used in [[iTunes]], Safari web browser, and other Apple products. Apple's AAC encoder is known to be one of the highest quality medium-bitrate [[CBR]] and [[VBR]] LC [[AAC]] encoders. The encoder started as part of the ''QuickTime'' media framework, but Apple has transitioned away from ''QuickTime'', to ''AV Foundation'' and ''AudioToolbox''.

The encoder does consistently well in [[Hydrogenaudio Listening Tests|Hydrogenaudio listening tests]].
__TOC__
== Bitrate modes ==
Four rate settings are present in Apple AAC:<ref>https://hydrogenaud.io/index.php/topic,118203.0.html Testing by Pr0m3th3u5 in 2019; comment by JAZ</ref>
; Constant bitrate (CBR)
: Similar to MP3 CBR in how it handles the bit reservior, but can still swing a little from the target
; Average bit rate (ABR)
: Self-explanatory
; Constrained VBR
: Like ABR, but seems to be more open to using more bits
; Variable bitrate (VBR)
; "True" VBR (TVBR, unofficial name)
: Takes a quality from 0 to 127 (kAudioCodecPropertySoundQualityForVBR). Seems to perform similarly to cVBR once the below table is used for bitrate conversion.

{|class=wikitable
|+TVBR bitrate mapping (AAC)
|-
! -ue / -Q !! ffmpeg q !! bitrate (bps)
|-
| 45 || 9.11 || 96
|-
| 64 || 7 || 128
|-
| 73 || 6 || 144
|-
| 82 || 5 || 160
|-
| 91 || 4 || 192
|-
| 109 || 2 || 256
|-
| 127 || 0 || 320
|}

The above is approximated by a piece-wise function, although the neatness is likely more indicative of how HA users pick values for tests:

Bitrate(q) = (40 * q - 1184) / 9 82 <= q < 128
(16 * q + 128) / 9 64 <= q < 82

TVBR is not available for HE-AAC.

== Format support ==
Apple's CoreAudioBaseTypes lists the following AAC-derived codecs on macOS 11.3:<ref name=CABT>https://github.com/phracker/MacOSX-SDKs/blob/041600eda65c6a668f66cb7d56b7d1da3e8bcc93/MacOSX11.3.sdk/System/Library/Frameworks/CoreAudioTypes.framework/Versions/A/Headers/CoreAudioBaseTypes.h#LL391C2-L391C</ref>
* AAC-LC (MPEG-4)
* AAC-LD
* AAC-ELD
* AAC-ELD + SBR
* HE-AAC v1 (AAC + SBR)
* HE-AAC v2 (v1 + PS)
* USAC
* MPEG-4 Spatial

There should also be support for a number of other format, at least in decoding: Apple does not ship a standalone AAC codec, but a whole audio-processing kit.

Actual support depends on library version and whether the program used to call the library knows how to use these formats.
* afconvert is clumsy but exposes almost everything.
* QAAC only does LC and HEv1. It can use some of the audiotoolkit filters while it works.
* FFmpeg only knows about using Apple AAC with LC, LD, ELD, HE, and HEv2.

== afconvert ==
Apple's macOS includes a command-line utility for transcoding audio files that employs this encoder when encoding to AAC.

Usage:
afconvert [options] <infile> <outfile>

Options:
;-f --file <string>:File format. Several; use 'm4af' for '.m4a' file or 'adts' for raw '.aac' file.
;-d --data <string>:Data format. Several; use 'aac' for AAC-LC. See CoreAudioBaseTypes.h link in footnote.<ref name=CABT/>
;-s --strategy <0,1,2,3>:Bitrate allocation strategy. 0 for CBR, 1 for ABR, 2 for VBR_constrained, 3 for VBR
;-b --bitrate <n>:Bitrate in bits per second. Does nothing when used with strategy 3.
;-ue vbrq <0-127>:Output quality for VBR mode. Only applicable when using strategy 3.
;-q --quality <0-127>:Speed/quality trade-off. Internally rounded to a value of either 32, 64, or 96.

afconvert has way more options than those listed above, but not all are useful for AAC. Run <code>afconvert --help</code> to see them all, or see [https://ss64.com/osx/afconvert.html SS64] for an unofficial list.

== QAAC ==
{{Software Infobox
| name = qaac
| logo =
| screenshot =
| caption =
| maintainer = nu774
| stable_release = 2.83
| preview_release =
| operating_system = Windows
| use = Encoder
| license = Various
| website = [https://github.com/nu774/qaac github/qaac]
}}

QAAC is an open-source wrapper for ''Core Audio's'' AAC and [[ALAC]] encoders that allow them to be used directly in applications that don't otherwise use ''Core Audio''. It exists only for Windows.

* [https://github.com/nu774/qaac/wiki QAAC documentation]

Linux users [https://www.andrews-corner.org/qaac.html should be able to use wine]. The most fragile part apparently is [https://github.com/enzo1982/freac/issues/168 getting AppleApplicationSupport to install].

 

== ffmpeg ==
On macOS, ffmpeg 4.4 and above can use the <code>aac_at</code> codec to use Apple AAC via the new AudioToolbox.framework. In addition to AAC-LC, HE-AAC (v1/v2), AAC-LD, and AAC-ELD are available, the same selection of profiles as FDK AAC. FFmpeg uses a 0-14 quality range to be converted to the 0-127 scale, using the expression <code>q = 127 - q * 9;</code>.<ref>https://github.com/FFmpeg/FFmpeg/blob/f009f849617ab0c2015d2a748892d55b37d378c0/libavcodec/audiotoolboxenc.c#L328</ref>

On Windows, ffmpeg's build process can be tricked into linking to an CoreAudioToolbox.dll commonly distributed alongside iTunes, using [https://github.com/dantmnf/AudioToolboxWrapper AudioToolboxWrapper].

==External links==
<references />
*[http://www.apple.com/quicktime Apple's Quicktime Website]

Foobar2000:Foobar2000

2024-06-11T18:44:12Z

Logan589XP: Updated stable release version number and release date.

{{title|foobar2000}}
{{Software Infobox|
|name = foobar2000
|logo = [[Image:foobar2000 Logo.png|48px]]
|screenshot = [[Image:Foobar2000-1.0-default-ui.png|250px]]
|caption = Screenshot of foobar2000 v1.0 using the default user interface
|developer = Peter Pawlowski
|released = {{start date and age|2002|12|20}}<ref>{{ha|https://hydrogenaud.io/index.php/topic,4997.0.html|Official Foobar2000 site & Foobar2000 0.3 & SDK!}}</ref>
|stable_release = 2.1.5<ref>{{foobar2000|http://www.foobar2000.org/download|Download foobar2000 for Windows}}</ref>
|stable_release_date = 2024-05-04
|preview_release =
|preview_release_date =
|operating_system = Windows
|use = Media Player
|license = Proprietary, BSD
|website = [http://www.foobar2000.org/ foobar2000.org]
}}
'''foobar2000''' is an advanced freeware audio player for the Windows platform. Some of the basic features include full Unicode support, [[ReplayGain]] support and native support for several popular audio formats.

== Platforms ==

foobar2000 has been written specifically for the Windows platform, and there are no plans to port it to any others. However, although not officially supported, it is known to run on [http://www.hydrogenaudio.org/forums/index.php?showtopic=54933 Linux] and [http://www.hydrogenaudio.org/forums/index.php?showtopic=77261 Mac OS X] through Wine and WineBottler, respectively. Also there is an [https://aur.archlinux.org/packages/foobar2000 Arch Linux native package] maintained by a community member.

== Features ==
* Powerful open component architecture allowing third-party developers to extend functionality of the player, including the ability to fully replace the user interface.
* Full Unicode support: File names, user interface, tagging, etc.
* [[ReplayGain]] support: Both playback and writing ReplayGain information to file tags.
* [[Gapless playback]].
* Advanced [[tagging]] capabilities - through built-in [[foobar2000:Properties|Properties dialog]] and various optional tagging-related components.
* Built-in [[foobar2000:Preferences:Media Library|Media Library]] functionality.
** Intuitive [[foobar2000:Query syntax|query syntax]] for searching the Media Library.
** Autoplaylist support: Generate dynamically updating playlists based on queries.
* [[foobar2000:Preferences:General:Keyboard Shortcuts|Customizable keyboard shortcuts]].
* Support for [[transcoding]] all supported audio formats using the [[Foobar2000:Converter|Converter component]] (requires external command-line encoder executables for different output formats).
* [[Secure_ripping|Secure]] [[foobar2000:Ripping CDs|CD ripping]].
* Streaming support.
* Efficient handling of large playlists.
* [[foobar2000:Components/Default_user_interface_%28foo_ui_std%29|User interface]] with simple configuration to create even complex layouts quickly and easily.
* Highly customizable display of track information using [[foobar2000:Titleformat_Introduction|title formatting scripts]].

==Supported audio formats==
Natively supported ("out-of-the-box"):
* [[MP1]], [[MP2]], [[MP3]], [[MP4]], [[Musepack]], [[AAC]], [[Ogg Vorbis]], [[FLAC]] / Ogg FLAC, [[Speex]], [[WavPack]], [[WAV]], [[AIFF]], [[AU|AU/SND]], [[CDDA]], [[WMA]], [[Matroska]], [[ALAC]], [[MMS]], [[RSTP]], [[Opus]], [[Monkey's Audio]], [[AC3]], [[DTS]], [[TAK]].
Supported through optional components:
* [[TTA]], [[MOD]], [[SPC]], [[Shorten]], [[OptimFROG]], [[PSF]], [[NSF]], [[XID]], [[XA]], [[AMR]], ''etc.''

In addition, foobar2000 can also play audio files within ZIP and RAR archives (typically compressed) directly, ''i.e.'' without requiring the user to extract the files first. More archive formats are supported ''via'' additional components: [http://www.foobar2000.org/components/view/foo_unpack_lha LHA/LZH], [http://www.foobar2000.org/components/view/foo_unpack_7z 7-Zip], [http://kode54.foobar2000.org/ JMA]

== Using foobar2000 ==

=== Important pages ===
* {{foobar2000|http://www.foobar2000.org/FAQ|foobar2000 FAQ}}
* [[foobar2000:components|foobar2000 components]]
* [[foobar2000:Legacy components|foobar2000 legacy components]]
* [[foobar2000:Encouraged Tag Standards|foobar2000 encouraged tag standards]]
* [[foobar2000:FAQ|foobar2000 FAQ (unofficial)]]
* [[:Category:Foobar2000_Guides|foobar2000 guides (category)]]
* [[:Category:Foobar2000_Preferences|foobar2000 preferences (category)]]

=== Specific guides ===

'''Preferences'''
* [[foobar2000:Preferences|Preferences dialog]]

'''Metadata'''
* [[foobar2000:Properties|Tag editing: the Properties dialog]]
* [[foobar2000:Query syntax|Query Syntax]]: details of Syntax for querying metadata.

'''Title formatting'''
* [[foobar2000:Title Formatting Introduction|Introduction to titleformat scripts]]
* [[foobar2000:Title Formatting Reference|Titleformat Reference]]: reference guide to all fields and functions
* [[foobar2000:Titleformat Examples|Titleformat Examples]]: user-submitted code for various purposes; submit your own!

'''Others'''
* [[foobar2000:File operations|File operations dialog]]: move, copy, rename, and delete files from within foobar2000
* [[foobar2000:Commandline Guide|Commandline usage]]

=== External Guides ===
* [http://www.audiohq.de/viewforum.php?id=27 German-language guides]
* [http://foobar2000.xrea.jp/ fb2k Wiki Page] for Japanese users.
* [http://winamp2foobar.blogspot.com Winamp To Foobar Guide] with information relevant for general users also.

=== Technical information ===
* [[foobar2000:ID3 Tag Mapping|ID3 Tag Mapping]]
* [[foobar2000:Metadata Compatibility|Metadata Compatibility]]: information about compatibility with metadata written by other applications

== Important Links ==
=== Official site ===
* [http://www.foobar2000.org foobar2000.org: Homepage]
* [http://www.foobar2000.org/download foobar2000.org: Download]
* [http://www.foobar2000.org/components foobar2000.org: Components]

=== Community ===
* Official IRC Channel: #foobar2000 on [https://libera.chat/ Libera Chat]
* {{ha|https://hydrogenaud.io/index.php/board,28.0.html|foobar2000 Forums}}
* [http://foobar-users.de/ German Support Forum]
* [http://www.fforum.ru/viewforum.php?f=59 Russian-language forum]
* [http://www.foobar2000.ru/forum/ Another Russian-language forum]

=== Appearance ===
* {{ha|https://hydrogenaud.io/index.php/topic,61333.0.html|Default UI .fth thread}} (fast way to clone another's DUI configuration)
* [[foobar2000:Preferences:Columns UI/Appearance|Columns UI appearance customization guides]]
* {{ha|https://hydrogenaud.io/index.php/topic,31027.0.html|Columns UI configurations}}

== References ==
<references/>

[[Category:foobar2000]]
[[Category:Media Players|foobar2000]]
[[Category:CD Rippers]]
[[Category:Software]]
[[Category:Tag editors]]

Opus

2024-04-24T05:26:01Z

Logan589XP: Updated stable release number to 1.5.2.

{{Software Infobox
| name = Opus
| logo = [[Image:opus-logo.png|250px|Official Opus logo]]
| screenshot =
| caption = Opus Interactive Audio Codec
| maintainer = [http://xiph.org/ Xiph.Org Foundation]
| stable_release = 1.5.2
| operating_system = Windows, Mac OS/X, Linux/BSD
| use = Encoder/Decoder
| license = 3-clause BSD license
| website = [http://www.opus-codec.org/ opus-codec.org]
}}

'''Opus''' is a [[lossy]] audio compression format developed by the Internet Engineering Task Force (IETF) designed to be suitable for interactive real-time applications over the Internet,{{ref|homepage|a}} including music as well as speech, yet it is also very competitive for use as a storage and playback format, being a [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ class leader at around 64 kbps] and [http://listening-test.coresv.net/results.htm also at 96 kbps]. As an open format standardised through [http://tools.ietf.org/html/rfc6716 Request for Comments (RFC) 6716],{{ref|RFC|c}} a high quality reference implementation is provided under the 3-clause BSD license{{ref|homepage|a}} which compiles and runs on the vast majority of general purpose and embedded (fixed point) processors. Many Software patents which cover Opus are licensed under royalty-free terms.{{ref|FAQ|b}} Opus is also a Mandatory To Implement (MTI) codec for the upcoming WebRTC (Web Real Time Communication) specification of the World Wide Web Consortium (W3C).

Opus incorporates technology from two codecs, the speech-oriented SILK codec developed by Skype and the multi-purpose low-latency CELT codec developed by Xiph.org with significant changes to each to ensure they can work together.{{ref|RFC|c}} Opus can seamlessly transition among high and low bitrates, using a linear prediction codec (the SILK layer) at lower bitrates and a lapped transform codec (the CELT layer) at higher bitrates, as well as a hybrid of the two for a short overlap in which SILK encodes the 0–8 kHz spectrum and the CELT layer encodes only the frequencies above 8kHz.{{ref|RFC|c}} Opus has very low algorithmic delay (typ 22.5 ms) compared to popular music formats such as [[MP3]], [[Vorbis |Ogg Vorbis]], [[AAC | LC-AAC and HE-AAC]] (all over 100 ms), yet performs very competitively with them in terms of quality per bitrate, making it comparably viable as a storage & playback format. Also unlike Vorbis, Opus does not require the definition of large codebooks for each individual file, making it also preferable for short clips of audio, such as those often used by game developers, a field where patent-free Vorbis is commonly used.{{ref|RFC|c}}

Considerably more details of the history and potential applications for Opus are included in the ''Wikipedia'' page for '''[http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Opus (audio format)]'''

==Characteristics==
Opus supports bitrates from 6 kbps to 510 kbps for typical stereo audio sources (and a maximum of around 255 kbps per channel for multichannel audio), with the 'sweet spot' for music and general audio around 30 kbps (mono) and 40–100 kbps (stereo). It is intrinsically [[VBR | variable bitrate]], though constrained VBR and [[CBR | constant bitrate]] modes are possible where required. In the case of the reference release, libopus, the target bitrate is calibrated against the internal constant quality targets so that over a typical music collection, something very close to the target bitrate will be achieved. This bitrate-calibrated approach differs from most VBR encoders (e.g. LAME, helix mp3, qaac, Nero aacenc, Ogg Vorbis, Musepack) where a setting on some 'constant quality' scale (which differs between encoders) is used and the bitrate will fall where it may. Improved future versions can be expected to offer improved quality at the same setting. Independent implementations may adopt a different approach.

Opus is able to seamlessly adapt its mode of operation without glitches or sound interruption (an illustrative demonstration of [http://opus-codec.org/examples/#gauge bitrate scalability] is on the Opus Examples page), which can be particularly useful for mixed-content audio or varying network conditions, making the unified Opus codec superior to a suite of different codecs that might otherwise cover the same range of bitrate and quality settings and would require out-of-band signalling to instigate codec switching. The switching includes the choice of mono, stereo and other channel mappings, the use of the speech-oriented SILK layer, the general-purpose CELT layer or the hybrid of both, and the use of different audio bandwidths (4, 6, 8, 12, or 20 kHz) as well as the quality adjustments within the same operating mode that are available in most VBR-capable codecs.

Of importance mainly to interactive uses, but potentially useful in time-delayed audio streaming also, Opus includes packet loss concealment (PLC) in all modes and, in the speech-oriented modes where the SILK layer is active it also supports Forward Error Correction (FEC) where the expected rate of packet loss can be indicated to the encoder by the user or by application software and critical frames (e.g. consonant sounds) can be retransmitted at low bitrate to preserve intelligibility.

For music and general audio, the CELT layer of Opus builds on knowledge gained during xiph.org's Vorbis development and ensures as a primary goal that the total energy in each spectral band is preserved while requiring only a modest bitrate overhead to achieve this, thereby eliminating a lot of bitrate-starvation artifacts such as 'birdies' that are common in low-bitrate MP3, especially during transients, applause and cymbal sounds. This technique likewise increases coding efficiency at bitrates targetting transparent music reproduction. Short blocks (2.5 ms) are also possible for efficient transient handling. Short blocks can also be used exclusively, if very low algorithmic delay (5.0 ms) is required to enable very low-latency interative audio (e.g. live networked music performances such as remote jam sessions), though greater bitrate is then required to maintain the same quality (illustrated in [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo Monty's CELT demo page] under Constant PEAQ value, varying latency). CELT uses a number of additional techniques and provides additional advanced tools to enable encoder tuning.

Opus natively supports [[gapless playback]] (though [[Gapless_playback#Poorly_designed_playback_systems | poor player design]] might itself induce interruptions during playback). Playback gain is also required, making some form of [[ReplayGain]] or [[ReplayGain_2.0_specification | similar]] volume control possible in any compliant player.

==Bitrate performance==
For mono speech, Opus ranges from intelligible narrowband speech reproduction starting at 6 kbps to medium-band, wideband and superwideband speech, reaching full-band speech by around 14 kbps in encoder version 1.2 (was 21 kbps in v1.1, 29 kbps in v1.0). Above about 32 kbps, the SILK layer is no longer used at all, as CELT alone gives superior quality.

For music, the SILK modes are quite tolerable and better than CELT at very low bitrates. The hybrid mode is adopted as bitrate increases, extending bandwidth first to 12 kHz (comparable with compact cassette) then to the full 20 kHz and CELT then takes over. Assuming the source is stereo, the transition from mono to stereo typically happens between the transition from 12 kHz to 20 kHz. Encoder version 1.2 includes great improvements to music encoding in the 32–64 kbps range, allowing full-band stereo at 32 kbps and providing acceptable quality at 48 kbps where artifacts are audible but rarely annoying. Version 1.3 is expected to further improve quality in this range.

Multi-format stereo music listening tests have demonstrated the superiority of Opus at 64 kbps and 96 kbps compared to the best AAC-LC, HE-AAC and Ogg Vorbis encoders, and at 96 kbps also to 128 kbps MP3 encoded using LAME <code>-V 5</code>.

==Indicative bitrate and quality==
The tables below give illustrative, indicative quality guidance based on typical modes used internally by Opus and a range of listening tests.

In encoder version 1.1 automatic detection of speech/music and bandwidth detection were introduced to improve mode decisions and VBR is less constrained, all with the aim of maximizing the quality/bitrate tradeoff, and these improvements are further enhanced in version 1.2 and 1.3. These tables are likely to require updates as the encoder is improved, especially in low-bitrate regions.

===Speech encoding quality===
This table assumes a '''monophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate) but mentions stereo compatibility for 40kbps+. The default 20ms frame size (22.5ms latency) is assumed. Note that the selection of ''VOIP'' mode will deliberately modify the sound with a High Pass Filter and emphasis of formants and harmonics to improve intelligibility of speech especially in noisy environments much as telephones do. ''Auto'' mode will not modify the sound prior to encoding so is usually better for high quality speech recordings or mixed speech and music.

{| class="wikitable" style="text-align:center"
|-
!Bitrate Target
!Bandwidth
!Typical Mode Used
!Speech Quality
!Use Cases / Competitive Codecs
|-
! Less than 6 kbps
| —
| —
| Bitrates lower than 6 kbps not supported by Opus (SILK disabled if forced to encode, which results in terrible speech quality)
| Try [https://en.wikipedia.org/wiki/Codec_2 Codec 2] for 0.45–3.2 kbps mono speech or [[Wikipedia:Lyra (codec)|Lyra]] for 3.2 kbps mono speech
|-
!6 kbps
|4 kHz narrow-band
|SILK
|Fair, intelligible
|AMR-NB may be a little better, but higher latency & proprietary, [[Speex]] also competitive
|-
!9 kbps VBR/CVBR 10 kbps CBR
|8 kHz wide-band
|SILK
|Telephone quality
|AMR-NB & AMR-WB similar quality, but higher latency & proprietary. [[Speex]] competitive.
|-
!12 kbps
|12 kHz super-wideband
|hybrid
|Medium bandwidth, better than telephone quality
|Similar quality to AMR-WB
|-
!16 kbps
|20 kHz
|hybrid/CELT
|Wideband speech quality
|Similar to/better than AMR-WB
|-
!24 kbps
|20 kHz
|hybrid/CELT
|Near transparent speech
|Better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!32 kbps
|20 kHz
|CELT
|Essentially transparent speech plus moderately good stereo music
|Much better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!40 kbps
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, fairly good stereo music
|Stereo podcasts/audiobooks/talk radio with some music
|-
!48 kbps or more
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, reasonable music
|Flexible general purpose modes to suit mixed music and speech
|-
|}

One major limitation of Opus at low bitrate is that SILK is inherently VBR: it accepts no constraints in CVBR, and if forced to do CBR the quality degrades from bit-shaving. As a result, even though constrained VBR is designed such that a fixed-rate data link requires at most one frame of buffer to handle the variation in bit rate -- great news for communication links -- any use of SILK, even in hybrid mode, has the potential of breaking this intention. This makes Opus suboptimal for low-rate radio links: radio links requires a predictable buffer amount, which is only possible with CBR when SILK is used, but use of CBR in turn hurts SILK. There is a noticeable quality difference at the NB/WB switch at 9 kbps VBR / 10 kbps CBR.

Opus 1.3+ allows forced use of SILK down to 5 kbps VBR (NB) and 6 kbps VBR (WB, requires forcing the C API with <code>OPUS_SET_BANDWIDTH</code>). However, quality is in no way guaranteed -- it's just possible.

===Music encoding quality===
This table assumes a '''stereophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate). Opus will automatically use mono at very low bitrates, though a certain amount of stereo encoding can still be used (content dependent) even when mono is specified as the typical stereo mode in the table below.

{| class="wikitable" style="text-align:center"
|-
!Bitrate target
!Stereo mode
!Bandwidth
!typ SILK/CELT use
!Music quality notes
!Use cases/notes/competitive codecs
|-
!4 kbps
|mono
|6 kHz
|SILK
|Poor, muffled sound but intelligible lyrics.
| —
|-
!9 kbps
|mono
|8 kHz
|SILK
|Poor, muffled but OK for bitrate
| —
|-
!14 to 16 kbps
|mono
|20 kHz
|hybrid/CELT
|Fairly poor but OK for bitrate
|Perhaps acceptable for incidental music
|-
!22 to 24 kbps
|mono
|20 kHz
|hybrid/CELT
|Fair but OK for bitrate
|OK for incidental music
|-
!32 to 40 kbps
|stereo
|20 kHz
|CELT
|Moderately good stereo, some artifacts, rarely nasty
|Stereo podcasts, audiobooks, very low bitrate music
|-
!48 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, may have problems with cymbals
|Stereo podcasts, audiobooks, low bitrate music
|-
!64 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, detectable differences to original (mostly 'not annoying')
|Music storage & streaming. Beat HE-AAC, Vorbis, MP3 in [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ listening test]
|-
!96 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, good quality approaching transparency
|Music storage & high quality streaming. Beat LC-AAC, Vorbis, MP3 in [http://listening-test.coresv.net/results.htm listening test]
|-
!112 kbps
|stereo
|20 kHz
|CELT
|Fairly close to transparency (needs more testing)
|Music storage & high quality streaming. Very low-latency stereo networked music performance/jam sessions at OK quality (see below table)
|-
!128 kbps
|stereo
|20 kHz
|CELT
|Very close to transparency (needs more testing). Most modern codecs competitive (AAC-LC, Vorbis, MP3)
|Music storage & streaming. Future download music sales.
|-
!160 to 192 kbps
|stereo
|20 kHz
|CELT
|Transparent with very low chance of artifacts (a few killer samples still detectable). Most old & new lossy codecs competitive.
|Music storage & streaming, dedicated limited-bandwidth audio links (e.g. wireless, [http://en.wikipedia.org/wiki/Bluetooth_profile#Advanced_Audio_Distribution_Profile_.28A2DP.29 A2DP-bluetooth] type links).
|-
!510 kbps
|stereo
|20 kHz
|CELT
|Maximum possible stereo bitrate target (actual rate often less than 510 for default frame size). Most old and new lossy codecs competitive, plus near-lossless [[lossyWAV]] and [[WavPack | WavPack lossy]]
|Music storage, dedicated limited-bitrate audio links (e.g. wireless, minimum latency high quality audio. LossyWAV and WavPack lossy are very competitive for storage, and WavPack lossy <code>--blocksize=256</code> may be competitive with minimum latency mode also.
|-
!>510 kbps
| —
| —
| —
|Above Opus bitrate range allowed for stereo sources
|Settle for 510 kbps or use [[lossless]], [[lossyWAV]], [[WavPack | WavPack lossy]] or lossy transform/subband codecs like [[Vorbis]], [[Musepack]] at very high settings.
|-
|}

===Lower latency versus quality/bitrate trade-off===
====Packet overhead in interactive applications====
For interactive use on the Internet or other packet-based networks, total bandwidth used will be subject to packet overhead. The more packet headers that are transmitted every second, the greater will be the overhead that is required. For this reason, Opus, while defaulting to 20 ms frames, supports 60 ms frames to reduce overhead when transporting low-bitrate SILK frames at the expense of greater latency, which may still be acceptable for speech, and also supports 10 ms SILK frames to reduce latency somewhat at the expense of packet overhead.

In the CELT layer, which tends to operate at higher bitrates than SILK, 20 ms frames are the default, but frames of 10 ms, 5 ms and 2.5 ms are also possible, which directly increases the frame overhead by transmitting more packets per second to achieve lower latency. In addition, as we'll see below it also reduces the quality/bitrate tradeoff of the CELT layer itself.

You probably do not want to use a frame size lower than 10 ms in applications containing speech, as doing so turns off SILK. The "lowdelay" application switch (available in FFmpeg and the raw library) turns off SILK to cut out 4 ms of synchronization delay, but a frame size of 10 ms achieves more delay reduction compared to default without sacrificing SILK.

None of the bitrates mentioned in this article account for the packet overhead.

====CELT layer latency versus quality/bitrate trade-off====
Unlike the SILK layer, which works on fixed 10 ms blocks, 1, 2 or 6 of which can be combined into an Opus frame, the CELT layer is able to modify the encoding block lengths available to enable its use with shorter frames.

When the CELT layer uses 10 ms, 5 ms and 2.5 ms frames instead of the default 20 ms, it must use smaller transform block sizes to achieve this, thereby reducing frequency resolution in the MDCT compared to the default transform window, thus reducing encoding efficiency for tonal signals. To obtain the same frequency precision for a sound divided into shorter transform windows, improved amplitude precision is necessary, resulting in increased bitrate to obtain the same perceptual quality (or conversely lower quality at the same bitrate).

These reduced-latency modes remain efficient for transient signals, which use short blocks anyway.

In all modes, the algorithmic delay consists of the frame size plus an additional 2.5 ms delay. The CELT layer requires 2.5 ms for MDCT window overlap.

Xiph.org used matched [[PEAQ]] scores (approximate perceptual quality assessment made in software) for the CELT0.10 codec that was used as the basis of the CELT layer in the Opus reference release, which indicate the following [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo approximate equivalent settings] for stereo music.

{| class="wikitable" style="text-align:center"
|-
!Frame size
!Algorithmic delay
!Bitrate to match 64kbps@22.5ms delay
!fractional bitrate increase
|-
!20 ms
|22.5 ms
|64.0 kbps
| +0.0 %
|-
!10 ms
|12.5 ms
|70.4 kbps
| +10.0 %
|-
!5 ms
|7.5 ms
|84.8 kbps
| +32.5 %
|-
!2.5 ms
|5.0 ms
|112.0 kbps
| +75.0 %
|-
|}

N.B. This table is useful for interactive streaming only. For music storage & delayed playback or non-interactive streaming, latency reduction is not important and the default 20 ms frame size is preferable.

==== "Equivalent bitrate" ====
Opus code includes a [https://github.com/xiph/opus/blob/9fc8fc4cf432640f284113ba502ee027268b0d9f/src/opus_encoder.c#L806 {{code|compute_equiv_rate()}}] function. Given the bitrate, framesize, cbr decision, and complexity setting, it converts the bitrate to an standard config (VBR, 20 ms frame, complexity 10) equivalent to be used for bandwidth, layer, and stereo decisions. The interesting bits are:

* CBR requires 8% more bitrate for the same quality.
* Frame overhead is fixed and modelled as <code>(40*channels+20)*(frame_rate - 50)</code> for any frame_rate larger than 50. (frame_rate is the number of frames per second, so <code>1000/frame_size_ms</code>). There's no modelling for reduction in overhead from larger-than-standard frames: you'd imagine the expression runs in the opposite direction as well.
* Complexity turning results in up to 30% more bitrate requirement.

This layer of conversion is why Opus runs wideband speech at 9 kbps VBR and CVBR, but with CBR it takes 10 kbps (now we know it's exactly 9.75 kbps) to use WB.

=== Channel count vs bitrate ===

For surround sound bitrates, use [[Bitrate#Equivalent bitrate estimates for multichannel audio]].

For ambisonics, see [https://www.mdpi.com/2076-3417/10/9/3188 AMBIQUAL listening test], paper figures 11 and 12.

== Implementations ==

The format and algorithms are openly documented and the reference implementation is published as free software. The reference implementation (Opus Audio Tools, opus-tools), consisting of separate encoders and decoders, is published under the terms of a BSD-like license. It is written in C programming language and can be compiled for hardware architectures with or without floating point unit. The accompanying diagnostic tool opusinfo reports detailed technical information about Opus files, including information on the standard compliance of the bitstream format. It is based on ogginfo from the vorbis-tools and therefore, unlike the encoder and decoder, available under the terms of version 2 of the GPL.

=== Reference implementation (libopus + binaries) ===
The commandline tools of the reference version are available pre-compiled for the most popular operating systems at [http://opus-codec.org/downloads opus-codec.org] and [https://ftp.mozilla.org/pub/mozilla.org/opus/ Mozilla's ftp server], plus in the foobar2000 free encoders pack and some alternative compiles through the hydrogenaud.io opus forum. The libopus commandline tools include encoder <code>opusenc</code>, decoder <code>opusdec</code>, and with a different license, the <code>opusinfo</code> opus stream & metadata analyzer.

The '''latest stable release''' is recommended for general use and as of mid 2014 is considered competitive with or superior to the best alternative speech or general music encoders at most supported bitrates.

==== libopus v1.0 ====
Released 11 Sep 2012 when RFC6716 was standardized but mostly fully developed by late 2011.

'''Stable, well-tuned''' <code>opusenc</code> reference encoder as included in RFC documentation.

CELT layer closely related to CELT 0.10 implements Constrained VBR mode by default (bitrate boost used mainly for transients), plus true CBR.

==== libopus v1.1 ====

The alpha source code released 21 Dec 2012 for testing & user feedback and following a beta release and testing, the stable 1.1 version was released on 5 December 2013, considered well tested enough for general release.<ref>https://people.xiph.org/~xiphmont/demo/opus/demo3.shtml</ref>

CELT layer [http://jmspeex.livejournal.com/11737.html quality improvements] introduced to provide '''unconstrained VBR''' include a rate boost not just for transients but now for highly tonal signals too and rate reduction when stereo image is narrow. There's also a rewrite of its '''transient detection''' code and '''time-frequency analysis''' code, and rewritten '''dynamic allocation''' code (HF/LF tilt and Band Boost) to allow more aggressive changes from the typical static allocation when warranted.

There are many minor improvements to '''speech quality''' in both SILK and CELT layers.

*'''DC-rejection''' below 3 Hz also aids quality if inaudible DC offset is present with no effect on deep bass notes.
*'''Automatic speech/music detection''' is introduced to optimize encoding mode choices, especially near the bitrate target range (presumably around 24–40 kbps) where the encoder may perform best with SILK, hybrid or CELT depending on content type. Below that range SILK performs best for both music & speech, and above it CELT performs best for speech & music. The detection, without look-ahead is not perfect but usually is undecided in audio where either mode will work well.
*'''Automatic bandwidth detection''' is also introduced to save wasted bits allocated to absent frequencies.
*'''Surround sound improvements''' were introduced since the beta release with considerable advances in coding efficiency, bitrate allocation and quality.

A new '''temporal VBR''' feature is added. For reasons not explained by classic psychoacoustics, it appears that giving more bits to loud frames (stealing from quiet frames) makes the result substantially better on listening tests. This feature is not tunable: it always affects VBR calculation at low bitrates, gradually becoming weaker at higher bitrates, until it turns off completely at 68 kbps.

==== libopus v1.1.3 ====
Released July 15th, 2016. This version contains:

*Neon optimizations improving performance on ARMv7 and ARMv8 by up to 15%
*Fixes some issues with 16-bit platforms (e.g. TI C55x)
*Fixes to comfort noise generation (CNG)
*Documenting that PLC packets can also be 2 bytes
*Includes experimental ambisonics work (<code>--enable-ambisonics</code>)

==== libopus v1.2.1 ====
Released June 26th, 2017. This version contains:

*Speech quality improvements especially in the 12–20 kbit/s range
*Improved VBR encoding for hybrid mode
*More aggressive use of wider speech bandwidth, including fullband speech starting at 14 kbit/s
*Music quality improvements in the 32–48 kbit/s range
*Generic and SSE CELT optimizations
*Support for directly encoding packets up to 120 ms
*DTX support for CELT mode
*SILK CBR improvements
*Support for all of the fixes in draft-ietf-codec-opus-update-06 (the mono downmix and the folding fixes need <code>--enable-update-draft</code>)
*Many bug fixes, including integer wrap-arounds discovered through fuzzing (no security implications)

==== libopus v1.3 ====
Released on October 18th, 2018. This version contains:

* Improvements to voice activity detection (VAD) and speech/music classification using a recurrent neural network (RNN)
* Support for ambisonics coding using channel mapping families 2 and 3
* Improvements to stereo speech coding at low bitrate
* Using wideband encoding down to 9 kb/s
* Making it possible to use SILK down to bitrates around 5 kb/s
* Minor quality improvement on tones
* Enabling the spec fixes in <nowiki>RFC 8251</nowiki> by default
* Security/hardening improvements
* Fixes to the CELT PLC
* Bandwidth detection fixes

==== libopus v1.3.1 ====
Released on April 12th, 2019. This version contains:

* Fixes to x87 builds
* A new OPUS_GET_IN_DTX query to know if the encoder is in DTX mode (last frame was either a comfort noise frame or not encoded at all)
* A new (and still experimental) CMake-based build system that is eventually meant to replace the VS2015 build system (the autotools one will stay)

==== libopus v1.4 ====
Released on April 20th, 2023. This version contains:

* Improved tuning of the Opus in-band FEC (LBRR). See the issue for details
* Added a OPUS_SET_INBAND_FEC(2) option that turns on FEC, but does not force SILK mode (FEC will be disabled in CELT mode)
* Improved tuning and various fixes to DTX
* Added Meson support, improved CMake support In addition to the improvements above, this release includes many minor bug fixes.

==== libopus v1.5.1 ====
Released on March 4th, 2024. This version contains:

* Significant improvement to packet loss robustness using Deep Redundancy (DRED)
* Improved packet loss concealment through Deep PLC
* Low-bitrate speech quality enhancement down to 6 kb/s wideband
* Improved x86 (AVX2) and Arm (Neon) optimizations
* Support for 4th and 5th order ambisonics

=== Other implementations ===

==== Concentus ====

The libopus reference library (fixed-point variant) has successfully been ported to both '''C#''' and '''Java''', as part of a project called '''Concentus'''. The aim of the project is specifically to target cross-platform applications where native C interop is relatively difficult. The code is available on [https://github.com/lostromb/concentus Github] and distributed via standard package managers.

==== Emscripten ports ====

At least one port of reference opus in Javascript has been made using the automated tool [https://developer.mozilla.org/en-US/docs/Mozilla/Projects/Emscripten emscripten]. See [https://blog.rillke.com/opusenc.js/ here], [https://github.com/kazuki/opus.js-sample here] and [https://github.com/audiocogs/opus.js here].

==== ffmpeg ====
FFmpeg has a native [https://ffmpeg.org/ffmpeg-codecs.html#opus "opus"] codec. It is of lower quality than the reference libopus and only does CELT coding. However, it is still good for the ecosystem to have a completely independent implementation.

== Hardware & Software Support ==

Much of this section is based heavily on the Jan 12th 2013 version of the '''Support''' section of the [http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Wikipedia article], which is more likely to be kept updated and to provide links to further information about the supporting platforms.

=== VoIP software ===
* The open source virtual PBX Freeswitch supports Opus transcoding.
* The voice-chat software Mumble supports Opus as its main codec.
* SIP softphones Phoner and PhonerLite support Opus
* The SIP and IAX2 client SFLphone is being fitted with Opus support.
* Integration of Opus into the Skype client is finished, although no version with Opus support has yet been published.
* TrueConf video conferencing solutions support Opus.
* Opus support is planned for Jitsi 2.0, together with VP8 video.
* Empathy may use any format supported in GStreamer, including Opus.
* Line2 has replaced their current codec with Opus. Their iOS app will be the first to be released with the Opus. The Android app will follow later.
* CSipSimple supports Opus, Codec2, G.726 and G.722.1 with an additional plug-in.
* The voice-chat software TeamSpeak 3 supports Opus for voice and music in pre-release server 3.0.7-pre2 and beta client version 3.0.10.
* The proprietary instant messenger service Discord uses Opus audio for all voice calls and video calls, regardless of platform.

=== Web frameworks and browsers ===
* Opus support is mandatory for WebRTC implementations.
* Mozilla supports Opus beginning with version 15 of Firefox and Thunderbird, plus Seamonkey, which uses a shared codebase.
* Depending on the backend in use, Opera supports inline playback of embedded Opus files. Official support for Opus and WebRTC are on the development roadmap.
* Chromium and Google Chrome have audio support as of version 33.
* Apple's Safari browser now supports Opus as of iOS 11 and macOS 10.13 High Sierra.
* Maxthon Cloud Browser

=== Streaming audio ===
* Icecast. (examples: [http://dir.xiph.org/by_format/Opus Stream directory by format Opus], [http://smj.delfa.net/opus_64.m3u 64k]/[http://smj.delfa.net/opus_256.m3u 256k] [http://smj.delfa.net/ Smooth Jazz Opus Stream], [http://www.absoluteradio.co.uk/listen/labs.html Absolute Radio Opus Trial] 7 stations at 24,64,96 kbps, [http://icecast.ofdoom.com:8000/burst-opus.ogg Icecast Of Doom 96k]
* Krad Radio
* Liquidsoap

=== Operating systems and desktop multimedia frameworks ===
* In Debian GNU/Linux the Opus development tools and supporting libraries can be installed from the preconfigured repositories in the next stable version ("wheezy") that is expected to be released in early 2013.
* For Microsoft Windows, there are DirectShow filters supporting Opus, including DC-Bass Source Mod and the LAV Filters.
* In GStreamer the integration of Opus support is complete.
* FFmpeg supports decoding and encoding Opus via the external library libopus.
* Android 5.0 and above supports Opus natively if encapsulated in the Ogg container, but .opus filename extension is not recognized by Android, so the use of double filename extension .opus.ogg is recommended as a workaround to allow apps to recognize files as playable audio.

=== Hardware support ===
* Support in [[Rockbox]] is available. This means hardware support for a series of portable media players (including some products from the iPod series by Apple and Sansa, iriver and Archos devices) and with "Rockbox as an Application" (RaaA) also on Android devices.

=== Player software ===

* Windows/Mac/Linux (Cross-Platform)
*# [[VLC]] (media player supports Opus as of version 2.0.4
*# [[Amarok]] 2.8 has transcoding support for Opus codec if ffmpeg is compiled with support for the libopus library & support for playback of Opus encoded files if Amarok is compiled against TagLib (newer than V1.8)
*# Clementine has Opus support
*# Audacious player
*# [[MPD]] as of version 0.18 if compiled against libopus (supports both encoding for http streams and decoding)
* Windows Exclusive
*# AIMP supports Opus natively as of version 3.20 build 1125 beta 1
*# [[foobar2000]] supports Opus natively as of v1.1.14 beta 1
*# Mpxplay supports Opus (using a decoder DLL) as of v1.60 alpha 2
*# [[Winamp]] supports Opus using a [http://forums.winamp.com/showthread.php?p=2925154#post2925154 3rd party plug-in]
*# MPC-HC
*# Resonic Player/Pro supports Opus natively as of version 0.2.2
* iOS/Android (Cross-Platform)
*# Capriccio [https://itunes.apple.com/us/app/capriccio-free-ultimate-music/id434829018?mt=8 iOS]/[https://play.google.com/store/apps/details?id=me.ideariboso.capriccio Android]
*# foobar2000 [https://itunes.apple.com/us/app/foobar2000/id1072807669?mt=8 iOS]/[https://play.google.com/store/apps/details?id=com.foobar2000.foobar2000&hl=en Android]
* Android Exclusive
*# [https://play.google.com/store/apps/details?id=in.krosbits.musicolet Musicolet Music Player]
*# [http://gonemadmusicplayer.blogspot.com/ GoneMAD Music Player]
*# [http://neutronmp.com/ Neutron Music Player]
*# [http://www.videolan.org/vlc/download-android.html VLC Media Player for Android]
*# [https://play.google.com/store/apps/details?id=ru.recoilme.freeamp FreeMP]
*# [https://play.google.com/store/apps/details?id=net.mderezynski.youki3 Youki]
*# [https://play.google.com/store/apps/details?id=com.aimp.player AIMP for Android]
*# [https://play.google.com/store/apps/details?id=com.acmeandroid.listen Listen Audiobook Player]
*# [https://play.google.com/store/apps/details?id=com.mxtech.videoplayer.ad MX Player]
*# [https://play.google.com/store/apps/details?id=org.tomahawk.tomahawk_android Tomahawk Player Beta]
*# [https://play.google.com/store/apps/details?id=com.maxmpz.audioplayer&hl=en Poweramp Music Player]

=== Other software ===
* CDBurnerXP
* MediaCoder
* Report-IT
* [[MP3tag|MP3tag]]
* [https://moisescardona.me/opus-gui/ Opus GUI]
* [http://www.xdlab.ru/en/ TagScanner]
* [http://www.xmedia-recode.de/ XMedia Recode]
* [[loudgain]]

== References & Notes ==

*{{note|homepage|a}}[http://opus-codec.org/ opus-codec.org homepage]
*{{note|FAQ|b}}[http://wiki.xiph.org/OpusFAQ Opus FAQ]
*{{note|RFC|c}}[http://tools.ietf.org/html/rfc6716 IETF RFC 6716]
<references/>
[[Category:Codecs]]
[[Category:Lossy]]
[[Category:Encoder/Decoder]]

Foobar2000:Free Encoder Pack

2024-04-24T04:49:26Z

Logan589XP: Updated encoder version numbers and release dates to reflect the newest Free Encoder Pack release.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2024-04-23
| stable_release_date = 2024-04-23
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.82
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.82
|
|-
| [[FLAC]]
| flac
| 1.4.3
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.5.2
|-
| [[WavPack]]
| wavpack
| 5.7.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Apple AAC

2024-03-31T03:30:37Z

Logan589XP: Updated version number to 2.82.

{{aac-encoders}}
'''Apple AAC''' is an [[AAC]] codec developed by Apple Inc. and included in Apple's audio framework, ''Core Audio''. ''Core Audio'' is used in [[iTunes]], Safari web browser, and other Apple products. Apple's AAC encoder is known to be one of the highest quality medium-bitrate [[CBR]] and [[VBR]] LC [[AAC]] encoders. The encoder started as part of the ''QuickTime'' media framework, but Apple has transitioned away from ''QuickTime'', to ''AV Foundation'' and ''AudioToolbox''.

The encoder does consistently well in [[Hydrogenaudio Listening Tests|Hydrogenaudio listening tests]].
__TOC__
== Bitrate modes ==
Four rate settings are present in Apple AAC:<ref>https://hydrogenaud.io/index.php/topic,118203.0.html Testing by Pr0m3th3u5 in 2019; comment by JAZ</ref>
; Constant bitrate (CBR)
: Similar to MP3 CBR in how it handles the bit reservior, but can still swing a little from the target
; Average bit rate (ABR)
: Self-explanatory
; Constrained VBR
: Like ABR, but seems to be more open to using more bits
; Variable bitrate (VBR)
; "True" VBR (TVBR, unofficial name)
: Takes a quality from 0 to 127 (kAudioCodecPropertySoundQualityForVBR). Seems to perform similarly to cVBR once the below table is used for bitrate conversion.

{|class=wikitable
|+TVBR bitrate mapping (AAC)
|-
! -ue / -Q !! ffmpeg q !! bitrate (bps)
|-
| 45 || 9.11 || 96
|-
| 64 || 7 || 128
|-
| 73 || 6 || 144
|-
| 82 || 5 || 160
|-
| 91 || 4 || 192
|-
| 109 || 2 || 256
|-
| 127 || 0 || 320
|}

The above is approximated by a piece-wise function, although the neatness is likely more indicative of how HA users pick values for tests:

Bitrate(q) = (40 * q - 1184) / 9 82 <= q < 128
(16 * q + 128) / 9 64 <= q < 82

TVBR is not available for HE-AAC.

== Format support ==
Apple's CoreAudioBaseTypes lists the following AAC-derived codecs on macOS 11.3:<ref name=CABT>https://github.com/phracker/MacOSX-SDKs/blob/041600eda65c6a668f66cb7d56b7d1da3e8bcc93/MacOSX11.3.sdk/System/Library/Frameworks/CoreAudioTypes.framework/Versions/A/Headers/CoreAudioBaseTypes.h#LL391C2-L391C</ref>
* AAC-LC (MPEG-4)
* AAC-LD
* AAC-ELD
* AAC-ELD + SBR
* HE-AAC v1 (AAC + SBR)
* HE-AAC v2 (v1 + PS)
* USAC
* MPEG-4 Spatial

There should also be support for a number of other format, at least in decoding: Apple does not ship a standalone AAC codec, but a whole audio-processing kit.

Actual support depends on library version and whether the program used to call the library knows how to use these formats.
* afconvert is clumsy but exposes almost everything.
* QAAC only does LC and HEv1. It can use some of the audiotoolkit filters while it works.
* FFmpeg only knows about using Apple AAC with LC, LD, ELD, HE, and HEv2.

== afconvert ==
Apple's macOS includes a command-line utility for transcoding audio files that employs this encoder when encoding to AAC.

Usage:
afconvert [options] <infile> <outfile>

Options:
;-f --file <string>:File format. Several; use 'm4af' for '.m4a' file or 'adts' for raw '.aac' file.
;-d --data <string>:Data format. Several; use 'aac' for AAC-LC. See CoreAudioBaseTypes.h link in footnote.<ref name=CABT/>
;-s --strategy <0,1,2,3>:Bitrate allocation strategy. 0 for CBR, 1 for ABR, 2 for VBR_constrained, 3 for VBR
;-b --bitrate <n>:Bitrate in bits per second. Does nothing when used with strategy 3.
;-ue vbrq <0-127>:Output quality for VBR mode. Only applicable when using strategy 3.
;-q --quality <0-127>:Speed/quality trade-off. Internally rounded to a value of either 32, 64, or 96.

afconvert has way more options than those listed above, but not all are useful for AAC. Run <code>afconvert --help</code> to see them all, or see [https://ss64.com/osx/afconvert.html SS64] for an unofficial list.

== QAAC ==
{{Software Infobox
| name = qaac
| logo =
| screenshot =
| caption =
| maintainer = nu774
| stable_release = 2.82
| preview_release =
| operating_system = Windows
| use = Encoder
| license = Various
| website = [https://github.com/nu774/qaac github/qaac]
}}

QAAC is an open-source wrapper for ''Core Audio's'' AAC and [[ALAC]] encoders that allow them to be used directly in applications that don't otherwise use ''Core Audio''. It exists only for Windows.

* [https://github.com/nu774/qaac/wiki QAAC documentation]

Linux users [https://www.andrews-corner.org/qaac.html should be able to use wine]. The most fragile part apparently is [https://github.com/enzo1982/freac/issues/168 getting AppleApplicationSupport to install].

 

== ffmpeg ==
On macOS, ffmpeg 4.4 and above can use the <code>aac_at</code> codec to use Apple AAC via the new AudioToolbox.framework. In addition to AAC-LC, HE-AAC (v1/v2), AAC-LD, and AAC-ELD are available, the same selection of profiles as FDK AAC. FFmpeg uses a 0-14 quality range to be converted to the 0-127 scale, using the expression <code>q = 127 - q * 9;</code>.<ref>https://github.com/FFmpeg/FFmpeg/blob/f009f849617ab0c2015d2a748892d55b37d378c0/libavcodec/audiotoolboxenc.c#L328</ref>

On Windows, ffmpeg's build process can be tricked into linking to an CoreAudioToolbox.dll commonly distributed alongside iTunes, using [https://github.com/dantmnf/AudioToolboxWrapper AudioToolboxWrapper].

==External links==
<references />
*[http://www.apple.com/quicktime Apple's Quicktime Website]

Opus

2024-03-09T05:11:56Z

Logan589XP: Added release notes for libopus 1.5.1 and updated stable release number to 1.5.1.

{{Software Infobox
| name = Opus
| logo = [[Image:opus-logo.png|250px|Official Opus logo]]
| screenshot =
| caption = Opus Interactive Audio Codec
| maintainer = [http://xiph.org/ Xiph.Org Foundation]
| stable_release = 1.5.1
| operating_system = Windows, Mac OS/X, Linux/BSD
| use = Encoder/Decoder
| license = 3-clause BSD license
| website = [http://www.opus-codec.org/ opus-codec.org]
}}

'''Opus''' is a [[lossy]] audio compression format developed by the Internet Engineering Task Force (IETF) designed to be suitable for interactive real-time applications over the Internet,{{ref|homepage|a}} including music as well as speech, yet it is also very competitive for use as a storage and playback format, being a [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ class leader at around 64 kbps] and [http://listening-test.coresv.net/results.htm also at 96 kbps]. As an open format standardised through [http://tools.ietf.org/html/rfc6716 Request for Comments (RFC) 6716],{{ref|RFC|c}} a high quality reference implementation is provided under the 3-clause BSD license{{ref|homepage|a}} which compiles and runs on the vast majority of general purpose and embedded (fixed point) processors. Many Software patents which cover Opus are licensed under royalty-free terms.{{ref|FAQ|b}} Opus is also a Mandatory To Implement (MTI) codec for the upcoming WebRTC (Web Real Time Communication) specification of the World Wide Web Consortium (W3C).

Opus incorporates technology from two codecs, the speech-oriented SILK codec developed by Skype and the multi-purpose low-latency CELT codec developed by Xiph.org with significant changes to each to ensure they can work together.{{ref|RFC|c}} Opus can seamlessly transition among high and low bitrates, using a linear prediction codec (the SILK layer) at lower bitrates and a lapped transform codec (the CELT layer) at higher bitrates, as well as a hybrid of the two for a short overlap in which SILK encodes the 0–8 kHz spectrum and the CELT layer encodes only the frequencies above 8kHz.{{ref|RFC|c}} Opus has very low algorithmic delay (typ 22.5 ms) compared to popular music formats such as [[MP3]], [[Vorbis |Ogg Vorbis]], [[AAC | LC-AAC and HE-AAC]] (all over 100 ms), yet performs very competitively with them in terms of quality per bitrate, making it comparably viable as a storage & playback format. Also unlike Vorbis, Opus does not require the definition of large codebooks for each individual file, making it also preferable for short clips of audio, such as those often used by game developers, a field where patent-free Vorbis is commonly used.{{ref|RFC|c}}

Considerably more details of the history and potential applications for Opus are included in the ''Wikipedia'' page for '''[http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Opus (audio format)]'''

==Characteristics==
Opus supports bitrates from 6 kbps to 510 kbps for typical stereo audio sources (and a maximum of around 255 kbps per channel for multichannel audio), with the 'sweet spot' for music and general audio around 30 kbps (mono) and 40–100 kbps (stereo). It is intrinsically [[VBR | variable bitrate]], though constrained VBR and [[CBR | constant bitrate]] modes are possible where required. In the case of the reference release, libopus, the target bitrate is calibrated against the internal constant quality targets so that over a typical music collection, something very close to the target bitrate will be achieved. This bitrate-calibrated approach differs from most VBR encoders (e.g. LAME, helix mp3, qaac, Nero aacenc, Ogg Vorbis, Musepack) where a setting on some 'constant quality' scale (which differs between encoders) is used and the bitrate will fall where it may. Improved future versions can be expected to offer improved quality at the same setting. Independent implementations may adopt a different approach.

Opus is able to seamlessly adapt its mode of operation without glitches or sound interruption (an illustrative demonstration of [http://opus-codec.org/examples/#gauge bitrate scalability] is on the Opus Examples page), which can be particularly useful for mixed-content audio or varying network conditions, making the unified Opus codec superior to a suite of different codecs that might otherwise cover the same range of bitrate and quality settings and would require out-of-band signalling to instigate codec switching. The switching includes the choice of mono, stereo and other channel mappings, the use of the speech-oriented SILK layer, the general-purpose CELT layer or the hybrid of both, and the use of different audio bandwidths (4, 6, 8, 12, or 20 kHz) as well as the quality adjustments within the same operating mode that are available in most VBR-capable codecs.

Of importance mainly to interactive uses, but potentially useful in time-delayed audio streaming also, Opus includes packet loss concealment (PLC) in all modes and, in the speech-oriented modes where the SILK layer is active it also supports Forward Error Correction (FEC) where the expected rate of packet loss can be indicated to the encoder by the user or by application software and critical frames (e.g. consonant sounds) can be retransmitted at low bitrate to preserve intelligibility.

For music and general audio, the CELT layer of Opus builds on knowledge gained during xiph.org's Vorbis development and ensures as a primary goal that the total energy in each spectral band is preserved while requiring only a modest bitrate overhead to achieve this, thereby eliminating a lot of bitrate-starvation artifacts such as 'birdies' that are common in low-bitrate MP3, especially during transients, applause and cymbal sounds. This technique likewise increases coding efficiency at bitrates targetting transparent music reproduction. Short blocks (2.5 ms) are also possible for efficient transient handling. Short blocks can also be used exclusively, if very low algorithmic delay (5.0 ms) is required to enable very low-latency interative audio (e.g. live networked music performances such as remote jam sessions), though greater bitrate is then required to maintain the same quality (illustrated in [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo Monty's CELT demo page] under Constant PEAQ value, varying latency). CELT uses a number of additional techniques and provides additional advanced tools to enable encoder tuning.

Opus natively supports [[gapless playback]] (though [[Gapless_playback#Poorly_designed_playback_systems | poor player design]] might itself induce interruptions during playback). Playback gain is also required, making some form of [[ReplayGain]] or [[ReplayGain_2.0_specification | similar]] volume control possible in any compliant player.

==Bitrate performance==
For mono speech, Opus ranges from intelligible narrowband speech reproduction starting at 6 kbps to medium-band, wideband and superwideband speech, reaching full-band speech by around 14 kbps in encoder version 1.2 (was 21 kbps in v1.1, 29 kbps in v1.0). Above about 32 kbps, the SILK layer is no longer used at all, as CELT alone gives superior quality.

For music, the SILK modes are quite tolerable and better than CELT at very low bitrates. The hybrid mode is adopted as bitrate increases, extending bandwidth first to 12 kHz (comparable with compact cassette) then to the full 20 kHz and CELT then takes over. Assuming the source is stereo, the transition from mono to stereo typically happens between the transition from 12 kHz to 20 kHz. Encoder version 1.2 includes great improvements to music encoding in the 32–64 kbps range, allowing full-band stereo at 32 kbps and providing acceptable quality at 48 kbps where artifacts are audible but rarely annoying. Version 1.3 is expected to further improve quality in this range.

Multi-format stereo music listening tests have demonstrated the superiority of Opus at 64 kbps and 96 kbps compared to the best AAC-LC, HE-AAC and Ogg Vorbis encoders, and at 96 kbps also to 128 kbps MP3 encoded using LAME <code>-V 5</code>.

==Indicative bitrate and quality==
The tables below give illustrative, indicative quality guidance based on typical modes used internally by Opus and a range of listening tests.

In encoder version 1.1 automatic detection of speech/music and bandwidth detection were introduced to improve mode decisions and VBR is less constrained, all with the aim of maximizing the quality/bitrate tradeoff, and these improvements are further enhanced in version 1.2 and 1.3. These tables are likely to require updates as the encoder is improved, especially in low-bitrate regions.

===Speech encoding quality===
This table assumes a '''monophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate) but mentions stereo compatibility for 40kbps+. The default 20ms frame size (22.5ms latency) is assumed. Note that the selection of ''VOIP'' mode will deliberately modify the sound with a High Pass Filter and emphasis of formants and harmonics to improve intelligibility of speech especially in noisy environments much as telephones do. ''Auto'' mode will not modify the sound prior to encoding so is usually better for high quality speech recordings or mixed speech and music.

{| class="wikitable" style="text-align:center"
|-
!Bitrate Target
!Bandwidth
!Typical Mode Used
!Speech Quality
!Use Cases / Competitive Codecs
|-
! Less than 6 kbps
| —
| —
| Bitrates lower than 6 kbps not supported by Opus (SILK disabled if forced to encode, which results in terrible speech quality)
| Try [https://en.wikipedia.org/wiki/Codec_2 Codec 2] for 0.45–3.2 kbps mono speech or [[Wikipedia:Lyra (codec)|Lyra]] for 3.2 kbps mono speech
|-
!6 kbps
|4 kHz narrow-band
|SILK
|Fair, intelligible
|AMR-NB may be a little better, but higher latency & proprietary, [[Speex]] also competitive
|-
!9 kbps VBR/CVBR 10 kbps CBR
|8 kHz wide-band
|SILK
|Telephone quality
|AMR-NB & AMR-WB similar quality, but higher latency & proprietary. [[Speex]] competitive.
|-
!12 kbps
|12 kHz super-wideband
|hybrid
|Medium bandwidth, better than telephone quality
|Similar quality to AMR-WB
|-
!16 kbps
|20 kHz
|hybrid/CELT
|Wideband speech quality
|Similar to/better than AMR-WB
|-
!24 kbps
|20 kHz
|hybrid/CELT
|Near transparent speech
|Better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!32 kbps
|20 kHz
|CELT
|Essentially transparent speech plus moderately good stereo music
|Much better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!40 kbps
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, fairly good stereo music
|Stereo podcasts/audiobooks/talk radio with some music
|-
!48 kbps or more
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, reasonable music
|Flexible general purpose modes to suit mixed music and speech
|-
|}

One major limitation of Opus at low bitrate is that SILK is inherently VBR: it accepts no constraints in CVBR, and if forced to do CBR the quality degrades from bit-shaving. As a result, even though constrained VBR is designed such that a fixed-rate data link requires at most one frame of buffer to handle the variation in bit rate -- great news for communication links -- any use of SILK, even in hybrid mode, has the potential of breaking this intention. This makes Opus suboptimal for low-rate radio links: radio links requires a predictable buffer amount, which is only possible with CBR when SILK is used, but use of CBR in turn hurts SILK. There is a noticeable quality difference at the NB/WB switch at 9 kbps VBR / 10 kbps CBR.

Opus 1.3+ allows forced use of SILK down to 5 kbps VBR (NB) and 6 kbps VBR (WB, requires forcing the C API with <code>OPUS_SET_BANDWIDTH</code>). However, quality is in no way guaranteed -- it's just possible.

===Music encoding quality===
This table assumes a '''stereophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate). Opus will automatically use mono at very low bitrates, though a certain amount of stereo encoding can still be used (content dependent) even when mono is specified as the typical stereo mode in the table below.

{| class="wikitable" style="text-align:center"
|-
!Bitrate target
!Stereo mode
!Bandwidth
!typ SILK/CELT use
!Music quality notes
!Use cases/notes/competitive codecs
|-
!4 kbps
|mono
|6 kHz
|SILK
|Poor, muffled sound but intelligible lyrics.
| —
|-
!9 kbps
|mono
|8 kHz
|SILK
|Poor, muffled but OK for bitrate
| —
|-
!14 to 16 kbps
|mono
|20 kHz
|hybrid/CELT
|Fairly poor but OK for bitrate
|Perhaps acceptable for incidental music
|-
!22 to 24 kbps
|mono
|20 kHz
|hybrid/CELT
|Fair but OK for bitrate
|OK for incidental music
|-
!32 to 40 kbps
|stereo
|20 kHz
|CELT
|Moderately good stereo, some artifacts, rarely nasty
|Stereo podcasts, audiobooks, very low bitrate music
|-
!48 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, may have problems with cymbals
|Stereo podcasts, audiobooks, low bitrate music
|-
!64 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, detectable differences to original (mostly 'not annoying')
|Music storage & streaming. Beat HE-AAC, Vorbis, MP3 in [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ listening test]
|-
!96 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, good quality approaching transparency
|Music storage & high quality streaming. Beat LC-AAC, Vorbis, MP3 in [http://listening-test.coresv.net/results.htm listening test]
|-
!112 kbps
|stereo
|20 kHz
|CELT
|Fairly close to transparency (needs more testing)
|Music storage & high quality streaming. Very low-latency stereo networked music performance/jam sessions at OK quality (see below table)
|-
!128 kbps
|stereo
|20 kHz
|CELT
|Very close to transparency (needs more testing). Most modern codecs competitive (AAC-LC, Vorbis, MP3)
|Music storage & streaming. Future download music sales.
|-
!160 to 192 kbps
|stereo
|20 kHz
|CELT
|Transparent with very low chance of artifacts (a few killer samples still detectable). Most old & new lossy codecs competitive.
|Music storage & streaming, dedicated limited-bandwidth audio links (e.g. wireless, [http://en.wikipedia.org/wiki/Bluetooth_profile#Advanced_Audio_Distribution_Profile_.28A2DP.29 A2DP-bluetooth] type links).
|-
!510 kbps
|stereo
|20 kHz
|CELT
|Maximum possible stereo bitrate target (actual rate often less than 510 for default frame size). Most old and new lossy codecs competitive, plus near-lossless [[lossyWAV]] and [[WavPack | WavPack lossy]]
|Music storage, dedicated limited-bitrate audio links (e.g. wireless, minimum latency high quality audio. LossyWAV and WavPack lossy are very competitive for storage, and WavPack lossy <code>--blocksize=256</code> may be competitive with minimum latency mode also.
|-
!>510 kbps
| —
| —
| —
|Above Opus bitrate range allowed for stereo sources
|Settle for 510 kbps or use [[lossless]], [[lossyWAV]], [[WavPack | WavPack lossy]] or lossy transform/subband codecs like [[Vorbis]], [[Musepack]] at very high settings.
|-
|}

===Lower latency versus quality/bitrate trade-off===
====Packet overhead in interactive applications====
For interactive use on the Internet or other packet-based networks, total bandwidth used will be subject to packet overhead. The more packet headers that are transmitted every second, the greater will be the overhead that is required. For this reason, Opus, while defaulting to 20 ms frames, supports 60 ms frames to reduce overhead when transporting low-bitrate SILK frames at the expense of greater latency, which may still be acceptable for speech, and also supports 10 ms SILK frames to reduce latency somewhat at the expense of packet overhead.

In the CELT layer, which tends to operate at higher bitrates than SILK, 20 ms frames are the default, but frames of 10 ms, 5 ms and 2.5 ms are also possible, which directly increases the frame overhead by transmitting more packets per second to achieve lower latency. In addition, as we'll see below it also reduces the quality/bitrate tradeoff of the CELT layer itself.

You probably do not want to use a frame size lower than 10 ms in applications containing speech, as doing so turns off SILK. The "lowdelay" application switch (available in FFmpeg and the raw library) turns off SILK to cut out 4 ms of synchronization delay, but a frame size of 10 ms achieves more delay reduction compared to default without sacrificing SILK.

None of the bitrates mentioned in this article account for the packet overhead.

====CELT layer latency versus quality/bitrate trade-off====
Unlike the SILK layer, which works on fixed 10 ms blocks, 1, 2 or 6 of which can be combined into an Opus frame, the CELT layer is able to modify the encoding block lengths available to enable its use with shorter frames.

When the CELT layer uses 10 ms, 5 ms and 2.5 ms frames instead of the default 20 ms, it must use smaller transform block sizes to achieve this, thereby reducing frequency resolution in the MDCT compared to the default transform window, thus reducing encoding efficiency for tonal signals. To obtain the same frequency precision for a sound divided into shorter transform windows, improved amplitude precision is necessary, resulting in increased bitrate to obtain the same perceptual quality (or conversely lower quality at the same bitrate).

These reduced-latency modes remain efficient for transient signals, which use short blocks anyway.

In all modes, the algorithmic delay consists of the frame size plus an additional 2.5 ms delay. The CELT layer requires 2.5 ms for MDCT window overlap.

Xiph.org used matched [[PEAQ]] scores (approximate perceptual quality assessment made in software) for the CELT0.10 codec that was used as the basis of the CELT layer in the Opus reference release, which indicate the following [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo approximate equivalent settings] for stereo music.

{| class="wikitable" style="text-align:center"
|-
!Frame size
!Algorithmic delay
!Bitrate to match 64kbps@22.5ms delay
!fractional bitrate increase
|-
!20 ms
|22.5 ms
|64.0 kbps
| +0.0 %
|-
!10 ms
|12.5 ms
|70.4 kbps
| +10.0 %
|-
!5 ms
|7.5 ms
|84.8 kbps
| +32.5 %
|-
!2.5 ms
|5.0 ms
|112.0 kbps
| +75.0 %
|-
|}

N.B. This table is useful for interactive streaming only. For music storage & delayed playback or non-interactive streaming, latency reduction is not important and the default 20 ms frame size is preferable.

==== "Equivalent bitrate" ====
Opus code includes a [https://github.com/xiph/opus/blob/9fc8fc4cf432640f284113ba502ee027268b0d9f/src/opus_encoder.c#L806 {{code|compute_equiv_rate()}}] function. Given the bitrate, framesize, cbr decision, and complexity setting, it converts the bitrate to an standard config (VBR, 20 ms frame, complexity 10) equivalent to be used for bandwidth, layer, and stereo decisions. The interesting bits are:

* CBR requires 8% more bitrate for the same quality.
* Frame overhead is fixed and modelled as <code>(40*channels+20)*(frame_rate - 50)</code> for any frame_rate larger than 50. (frame_rate is the number of frames per second, so <code>1000/frame_size_ms</code>). There's no modelling for reduction in overhead from larger-than-standard frames: you'd imagine the expression runs in the opposite direction as well.
* Complexity turning results in up to 30% more bitrate requirement.

This layer of conversion is why Opus runs wideband speech at 9 kbps VBR and CVBR, but with CBR it takes 10 kbps (now we know it's exactly 9.75 kbps) to use WB.

=== Channel count vs bitrate ===

For surround sound bitrates, use [[Bitrate#Equivalent bitrate estimates for multichannel audio]].

For ambisonics, see [https://www.mdpi.com/2076-3417/10/9/3188 AMBIQUAL listening test], paper figures 11 and 12.

== Implementations ==

The format and algorithms are openly documented and the reference implementation is published as free software. The reference implementation (Opus Audio Tools, opus-tools), consisting of separate encoders and decoders, is published under the terms of a BSD-like license. It is written in C programming language and can be compiled for hardware architectures with or without floating point unit. The accompanying diagnostic tool opusinfo reports detailed technical information about Opus files, including information on the standard compliance of the bitstream format. It is based on ogginfo from the vorbis-tools and therefore, unlike the encoder and decoder, available under the terms of version 2 of the GPL.

=== Reference implementation (libopus + binaries) ===
The commandline tools of the reference version are available pre-compiled for the most popular operating systems at [http://opus-codec.org/downloads opus-codec.org] and [https://ftp.mozilla.org/pub/mozilla.org/opus/ Mozilla's ftp server], plus in the foobar2000 free encoders pack and some alternative compiles through the hydrogenaud.io opus forum. The libopus commandline tools include encoder <code>opusenc</code>, decoder <code>opusdec</code>, and with a different license, the <code>opusinfo</code> opus stream & metadata analyzer.

The '''latest stable release''' is recommended for general use and as of mid 2014 is considered competitive with or superior to the best alternative speech or general music encoders at most supported bitrates.

==== libopus v1.0 ====
Released 11 Sep 2012 when RFC6716 was standardized but mostly fully developed by late 2011.

'''Stable, well-tuned''' <code>opusenc</code> reference encoder as included in RFC documentation.

CELT layer closely related to CELT 0.10 implements Constrained VBR mode by default (bitrate boost used mainly for transients), plus true CBR.

==== libopus v1.1 ====

The alpha source code released 21 Dec 2012 for testing & user feedback and following a beta release and testing, the stable 1.1 version was released on 5 December 2013, considered well tested enough for general release.<ref>https://people.xiph.org/~xiphmont/demo/opus/demo3.shtml</ref>

CELT layer [http://jmspeex.livejournal.com/11737.html quality improvements] introduced to provide '''unconstrained VBR''' include a rate boost not just for transients but now for highly tonal signals too and rate reduction when stereo image is narrow. There's also a rewrite of its '''transient detection''' code and '''time-frequency analysis''' code, and rewritten '''dynamic allocation''' code (HF/LF tilt and Band Boost) to allow more aggressive changes from the typical static allocation when warranted.

There are many minor improvements to '''speech quality''' in both SILK and CELT layers.

*'''DC-rejection''' below 3 Hz also aids quality if inaudible DC offset is present with no effect on deep bass notes.
*'''Automatic speech/music detection''' is introduced to optimize encoding mode choices, especially near the bitrate target range (presumably around 24–40 kbps) where the encoder may perform best with SILK, hybrid or CELT depending on content type. Below that range SILK performs best for both music & speech, and above it CELT performs best for speech & music. The detection, without look-ahead is not perfect but usually is undecided in audio where either mode will work well.
*'''Automatic bandwidth detection''' is also introduced to save wasted bits allocated to absent frequencies.
*'''Surround sound improvements''' were introduced since the beta release with considerable advances in coding efficiency, bitrate allocation and quality.

A new '''temporal VBR''' feature is added. For reasons not explained by classic psychoacoustics, it appears that giving more bits to loud frames (stealing from quiet frames) makes the result substantially better on listening tests. This feature is not tunable: it always affects VBR calculation at low bitrates, gradually becoming weaker at higher bitrates, until it turns off completely at 68 kbps.

==== libopus v1.1.3 ====
Released July 15th, 2016. This version contains:

*Neon optimizations improving performance on ARMv7 and ARMv8 by up to 15%
*Fixes some issues with 16-bit platforms (e.g. TI C55x)
*Fixes to comfort noise generation (CNG)
*Documenting that PLC packets can also be 2 bytes
*Includes experimental ambisonics work (<code>--enable-ambisonics</code>)

==== libopus v1.2.1 ====
Released June 26th, 2017. This version contains:

*Speech quality improvements especially in the 12–20 kbit/s range
*Improved VBR encoding for hybrid mode
*More aggressive use of wider speech bandwidth, including fullband speech starting at 14 kbit/s
*Music quality improvements in the 32–48 kbit/s range
*Generic and SSE CELT optimizations
*Support for directly encoding packets up to 120 ms
*DTX support for CELT mode
*SILK CBR improvements
*Support for all of the fixes in draft-ietf-codec-opus-update-06 (the mono downmix and the folding fixes need <code>--enable-update-draft</code>)
*Many bug fixes, including integer wrap-arounds discovered through fuzzing (no security implications)

==== libopus v1.3 ====
Released on October 18th, 2018. This version contains:

* Improvements to voice activity detection (VAD) and speech/music classification using a recurrent neural network (RNN)
* Support for ambisonics coding using channel mapping families 2 and 3
* Improvements to stereo speech coding at low bitrate
* Using wideband encoding down to 9 kb/s
* Making it possible to use SILK down to bitrates around 5 kb/s
* Minor quality improvement on tones
* Enabling the spec fixes in <nowiki>RFC 8251</nowiki> by default
* Security/hardening improvements
* Fixes to the CELT PLC
* Bandwidth detection fixes

==== libopus v1.3.1 ====
Released on April 12th, 2019. This version contains:

* Fixes to x87 builds
* A new OPUS_GET_IN_DTX query to know if the encoder is in DTX mode (last frame was either a comfort noise frame or not encoded at all)
* A new (and still experimental) CMake-based build system that is eventually meant to replace the VS2015 build system (the autotools one will stay)

==== libopus v1.4 ====
Released on April 20th, 2023. This version contains:

* Improved tuning of the Opus in-band FEC (LBRR). See the issue for details
* Added a OPUS_SET_INBAND_FEC(2) option that turns on FEC, but does not force SILK mode (FEC will be disabled in CELT mode)
* Improved tuning and various fixes to DTX
* Added Meson support, improved CMake support In addition to the improvements above, this release includes many minor bug fixes.

==== libopus v1.5.1 ====
Released on March 4th, 2024. This version contains:

* Significant improvement to packet loss robustness using Deep Redundancy (DRED)
* Improved packet loss concealment through Deep PLC
* Low-bitrate speech quality enhancement down to 6 kb/s wideband
* Improved x86 (AVX2) and Arm (Neon) optimizations
* Support for 4th and 5th order ambisonics

=== Other implementations ===

==== Concentus ====

The libopus reference library (fixed-point variant) has successfully been ported to both '''C#''' and '''Java''', as part of a project called '''Concentus'''. The aim of the project is specifically to target cross-platform applications where native C interop is relatively difficult. The code is available on [https://github.com/lostromb/concentus Github] and distributed via standard package managers.

==== Emscripten ports ====

At least one port of reference opus in Javascript has been made using the automated tool [https://developer.mozilla.org/en-US/docs/Mozilla/Projects/Emscripten emscripten]. See [https://blog.rillke.com/opusenc.js/ here], [https://github.com/kazuki/opus.js-sample here] and [https://github.com/audiocogs/opus.js here].

==== ffmpeg ====
FFmpeg has a native [https://ffmpeg.org/ffmpeg-codecs.html#opus "opus"] codec. It is of lower quality than the reference libopus and only does CELT coding. However, it is still good for the ecosystem to have a completely independent implementation.

== Hardware & Software Support ==

Much of this section is based heavily on the Jan 12th 2013 version of the '''Support''' section of the [http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Wikipedia article], which is more likely to be kept updated and to provide links to further information about the supporting platforms.

=== VoIP software ===
* The open source virtual PBX Freeswitch supports Opus transcoding.
* The voice-chat software Mumble supports Opus as its main codec.
* SIP softphones Phoner and PhonerLite support Opus
* The SIP and IAX2 client SFLphone is being fitted with Opus support.
* Integration of Opus into the Skype client is finished, although no version with Opus support has yet been published.
* TrueConf video conferencing solutions support Opus.
* Opus support is planned for Jitsi 2.0, together with VP8 video.
* Empathy may use any format supported in GStreamer, including Opus.
* Line2 has replaced their current codec with Opus. Their iOS app will be the first to be released with the Opus. The Android app will follow later.
* CSipSimple supports Opus, Codec2, G.726 and G.722.1 with an additional plug-in.
* The voice-chat software TeamSpeak 3 supports Opus for voice and music in pre-release server 3.0.7-pre2 and beta client version 3.0.10.
* The proprietary instant messenger service Discord uses Opus audio for all voice calls and video calls, regardless of platform.

=== Web frameworks and browsers ===
* Opus support is mandatory for WebRTC implementations.
* Mozilla supports Opus beginning with version 15 of Firefox and Thunderbird, plus Seamonkey, which uses a shared codebase.
* Depending on the backend in use, Opera supports inline playback of embedded Opus files. Official support for Opus and WebRTC are on the development roadmap.
* Chromium and Google Chrome have audio support as of version 33.
* Apple's Safari browser now supports Opus as of iOS 11 and macOS 10.13 High Sierra.
* Maxthon Cloud Browser

=== Streaming audio ===
* Icecast. (examples: [http://dir.xiph.org/by_format/Opus Stream directory by format Opus], [http://smj.delfa.net/opus_64.m3u 64k]/[http://smj.delfa.net/opus_256.m3u 256k] [http://smj.delfa.net/ Smooth Jazz Opus Stream], [http://www.absoluteradio.co.uk/listen/labs.html Absolute Radio Opus Trial] 7 stations at 24,64,96 kbps, [http://icecast.ofdoom.com:8000/burst-opus.ogg Icecast Of Doom 96k]
* Krad Radio
* Liquidsoap

=== Operating systems and desktop multimedia frameworks ===
* In Debian GNU/Linux the Opus development tools and supporting libraries can be installed from the preconfigured repositories in the next stable version ("wheezy") that is expected to be released in early 2013.
* For Microsoft Windows, there are DirectShow filters supporting Opus, including DC-Bass Source Mod and the LAV Filters.
* In GStreamer the integration of Opus support is complete.
* FFmpeg supports decoding and encoding Opus via the external library libopus.
* Android 5.0 and above supports Opus natively if encapsulated in the Ogg container, but .opus filename extension is not recognized by Android, so the use of double filename extension .opus.ogg is recommended as a workaround to allow apps to recognize files as playable audio.

=== Hardware support ===
* Support in [[Rockbox]] is available. This means hardware support for a series of portable media players (including some products from the iPod series by Apple and Sansa, iriver and Archos devices) and with "Rockbox as an Application" (RaaA) also on Android devices.

=== Player software ===

* Windows/Mac/Linux (Cross-Platform)
*# [[VLC]] (media player supports Opus as of version 2.0.4
*# [[Amarok]] 2.8 has transcoding support for Opus codec if ffmpeg is compiled with support for the libopus library & support for playback of Opus encoded files if Amarok is compiled against TagLib (newer than V1.8)
*# Clementine has Opus support
*# Audacious player
*# [[MPD]] as of version 0.18 if compiled against libopus (supports both encoding for http streams and decoding)
* Windows Exclusive
*# AIMP supports Opus natively as of version 3.20 build 1125 beta 1
*# [[foobar2000]] supports Opus natively as of v1.1.14 beta 1
*# Mpxplay supports Opus (using a decoder DLL) as of v1.60 alpha 2
*# [[Winamp]] supports Opus using a [http://forums.winamp.com/showthread.php?p=2925154#post2925154 3rd party plug-in]
*# MPC-HC
*# Resonic Player/Pro supports Opus natively as of version 0.2.2
* iOS/Android (Cross-Platform)
*# Capriccio [https://itunes.apple.com/us/app/capriccio-free-ultimate-music/id434829018?mt=8 iOS]/[https://play.google.com/store/apps/details?id=me.ideariboso.capriccio Android]
*# foobar2000 [https://itunes.apple.com/us/app/foobar2000/id1072807669?mt=8 iOS]/[https://play.google.com/store/apps/details?id=com.foobar2000.foobar2000&hl=en Android]
* Android Exclusive
*# [https://play.google.com/store/apps/details?id=in.krosbits.musicolet Musicolet Music Player]
*# [http://gonemadmusicplayer.blogspot.com/ GoneMAD Music Player]
*# [http://neutronmp.com/ Neutron Music Player]
*# [http://www.videolan.org/vlc/download-android.html VLC Media Player for Android]
*# [https://play.google.com/store/apps/details?id=ru.recoilme.freeamp FreeMP]
*# [https://play.google.com/store/apps/details?id=net.mderezynski.youki3 Youki]
*# [https://play.google.com/store/apps/details?id=com.aimp.player AIMP for Android]
*# [https://play.google.com/store/apps/details?id=com.acmeandroid.listen Listen Audiobook Player]
*# [https://play.google.com/store/apps/details?id=com.mxtech.videoplayer.ad MX Player]
*# [https://play.google.com/store/apps/details?id=org.tomahawk.tomahawk_android Tomahawk Player Beta]
*# [https://play.google.com/store/apps/details?id=com.maxmpz.audioplayer&hl=en Poweramp Music Player]

=== Other software ===
* CDBurnerXP
* MediaCoder
* Report-IT
* [[MP3tag|MP3tag]]
* [https://moisescardona.me/opus-gui/ Opus GUI]
* [http://www.xdlab.ru/en/ TagScanner]
* [http://www.xmedia-recode.de/ XMedia Recode]
* [[loudgain]]

== References & Notes ==

*{{note|homepage|a}}[http://opus-codec.org/ opus-codec.org homepage]
*{{note|FAQ|b}}[http://wiki.xiph.org/OpusFAQ Opus FAQ]
*{{note|RFC|c}}[http://tools.ietf.org/html/rfc6716 IETF RFC 6716]
<references/>
[[Category:Codecs]]
[[Category:Lossy]]
[[Category:Encoder/Decoder]]

Foobar2000:Free Encoder Pack

2024-03-09T03:57:48Z

Logan589XP: Updated encoder version numbers and release dates to reflect the newest Free Encoder Pack release.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2024-03-08
| stable_release_date = 2024-03-08
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.81
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.81
|
|-
| [[FLAC]]
| flac
| 1.4.3
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.5.1
|-
| [[WavPack]]
| wavpack
| 5.7.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Apple AAC

2024-02-19T09:41:38Z

Logan589XP: Updated version number to 2.81.

{{aac-encoders}}
'''Apple AAC''' is an [[AAC]] codec developed by Apple Inc. and included in Apple's audio framework, ''Core Audio''. ''Core Audio'' is used in [[iTunes]], Safari web browser, and other Apple products. Apple's AAC encoder is known to be one of the highest quality medium-bitrate [[CBR]] and [[VBR]] LC [[AAC]] encoders. The encoder started as part of the ''QuickTime'' media framework, but Apple has transitioned away from ''QuickTime'', to ''AV Foundation'' and ''AudioToolbox''.

The encoder does consistently well in [[Hydrogenaudio Listening Tests|Hydrogenaudio listening tests]].
__TOC__
== Bitrate modes ==
Four rate settings are present in Apple AAC:<ref>https://hydrogenaud.io/index.php/topic,118203.0.html Testing by Pr0m3th3u5 in 2019; comment by JAZ</ref>
; Constant bitrate (CBR)
: Similar to MP3 CBR in how it handles the bit reservior, but can still swing a little from the target
; Average bit rate (ABR)
: Self-explanatory
; Constrained VBR
: Like ABR, but seems to be more open to using more bits
; Variable bitrate (VBR)
; "True" VBR (TVBR, unofficial name)
: Takes a quality from 0 to 127 (kAudioCodecPropertySoundQualityForVBR). Seems to perform similarly to cVBR once the below table is used for bitrate conversion.

{|class=wikitable
|+TVBR bitrate mapping (AAC)
|-
! -ue / -Q !! ffmpeg q !! bitrate (bps)
|-
| 45 || 9.11 || 96
|-
| 64 || 7 || 128
|-
| 73 || 6 || 144
|-
| 82 || 5 || 160
|-
| 91 || 4 || 192
|-
| 109 || 2 || 256
|-
| 127 || 0 || 320
|}

The above is approximated by a piece-wise function, although the neatness is likely more indicative of how HA users pick values for tests:

Bitrate(q) = (40 * q - 1184) / 9 82 <= q < 128
(16 * q + 128) / 9 64 <= q < 82

TVBR is not available for HE-AAC.

== Format support ==
Apple's CoreAudioBaseTypes lists the following AAC-derived codecs on macOS 11.3:<ref name=CABT>https://github.com/phracker/MacOSX-SDKs/blob/041600eda65c6a668f66cb7d56b7d1da3e8bcc93/MacOSX11.3.sdk/System/Library/Frameworks/CoreAudioTypes.framework/Versions/A/Headers/CoreAudioBaseTypes.h#LL391C2-L391C</ref>
* AAC-LC (MPEG-4)
* AAC-LD
* AAC-ELD
* AAC-ELD + SBR
* HE-AAC v1 (AAC + SBR)
* HE-AAC v2 (v1 + PS)
* USAC
* MPEG-4 Spatial

There should also be support for a number of other format, at least in decoding: Apple does not ship a standalone AAC codec, but a whole audio-processing kit.

Actual support depends on library version and whether the program used to call the library knows how to use these formats.
* afconvert is clumsy but exposes almost everything.
* QAAC only does LC and HEv1. It can use some of the audiotoolkit filters while it works.
* FFmpeg only knows about using Apple AAC with LC, LD, ELD, HE, and HEv2.

== afconvert ==
Apple's macOS includes a command-line utility for transcoding audio files that employs this encoder when encoding to AAC.

Usage:
afconvert [options] <infile> <outfile>

Options:
;-f --file <string>:File format. Several; use 'm4af' for '.m4a' file or 'adts' for raw '.aac' file.
;-d --data <string>:Data format. Several; use 'aac' for AAC-LC. See CoreAudioBaseTypes.h link in footnote.<ref name=CABT/>
;-s --strategy <0,1,2,3>:Bitrate allocation strategy. 0 for CBR, 1 for ABR, 2 for VBR_constrained, 3 for VBR
;-b --bitrate <n>:Bitrate in bits per second. Does nothing when used with strategy 3.
;-ue vbrq <0-127>:Output quality for VBR mode. Only applicable when using strategy 3.
;-q --quality <0-127>:Speed/quality trade-off. Internally rounded to a value of either 32, 64, or 96.

afconvert has way more options than those listed above, but not all are useful for AAC. Run <code>afconvert --help</code> to see them all, or see [https://ss64.com/osx/afconvert.html SS64] for an unofficial list.

== QAAC ==
{{Software Infobox
| name = qaac
| logo =
| screenshot =
| caption =
| maintainer = nu774
| stable_release = 2.81
| preview_release =
| operating_system = Windows
| use = Encoder
| license = Various
| website = [https://github.com/nu774/qaac github/qaac]
}}

QAAC is an open-source wrapper for ''Core Audio's'' AAC and [[ALAC]] encoders that allow them to be used directly in applications that don't otherwise use ''Core Audio''. It exists only for Windows.

* [https://github.com/nu774/qaac/wiki QAAC documentation]

Linux users [https://www.andrews-corner.org/qaac.html should be able to use wine]. The most fragile part apparently is [https://github.com/enzo1982/freac/issues/168 getting AppleApplicationSupport to install].

 

== ffmpeg ==
On macOS, ffmpeg 4.4 and above can use the <code>aac_at</code> codec to use Apple AAC via the new AudioToolbox.framework. In addition to AAC-LC, HE-AAC (v1/v2), AAC-LD, and AAC-ELD are available, the same selection of profiles as FDK AAC. FFmpeg uses a 0-14 quality range to be converted to the 0-127 scale, using the expression <code>q = 127 - q * 9;</code>.<ref>https://github.com/FFmpeg/FFmpeg/blob/f009f849617ab0c2015d2a748892d55b37d378c0/libavcodec/audiotoolboxenc.c#L328</ref>

On Windows, ffmpeg's build process can be tricked into linking to an CoreAudioToolbox.dll commonly distributed alongside iTunes, using [https://github.com/dantmnf/AudioToolboxWrapper AudioToolboxWrapper].

==External links==
<references />
*[http://www.apple.com/quicktime Apple's Quicktime Website]

Apple AAC

2023-07-06T03:28:09Z

Logan589XP: Updated version number to 2.80.

{{aac-encoders}}
'''Apple AAC''' is an [[AAC]] codec developed by Apple Inc. and included in Apple's audio framework, ''Core Audio''. ''Core Audio'' is used in [[iTunes]], Safari web browser, and other Apple products. Apple's AAC encoder is known to be one of the highest quality medium-bitrate [[CBR]] and [[VBR]] LC [[AAC]] encoders. The encoder started as part of the ''QuickTime'' media framework, but Apple has transitioned away from ''QuickTime'', to ''AV Foundation'' and ''AudioToolbox''.

The encoder does consistently well in [[Hydrogenaudio Listening Tests|Hydrogenaudio listening tests]].
__TOC__
== Bitrate modes ==
Four rate settings are present in Apple AAC:<ref>https://hydrogenaud.io/index.php/topic,118203.0.html Testing by Pr0m3th3u5 in 2019; comment by JAZ</ref>
; Constant bitrate (CBR)
: Similar to MP3 CBR in how it handles the bit reservior, but can still swing a little from the target
; Average bit rate (ABR)
: Self-explanatory
; Constrained VBR
: Like ABR, but seems to be more open to using more bits
; Variable bitrate (VBR)
; "True" VBR (TVBR, unofficial name)
: Takes a quality from 0 to 127 (kAudioCodecPropertySoundQualityForVBR). Seems to perform similarly to cVBR once the below table is used for bitrate conversion.

{|class=wikitable
|+TVBR bitrate mapping (AAC)
|-
! -ue / -Q !! ffmpeg q !! bitrate (bps)
|-
| 45 || 9.11 || 96
|-
| 64 || 7 || 128
|-
| 73 || 6 || 144
|-
| 82 || 5 || 160
|-
| 91 || 4 || 192
|-
| 109 || 2 || 256
|-
| 127 || 0 || 320
|}

The above is approximated by a piece-wise function, although the neatness is likely more indicative of how HA users pick values for tests:

Bitrate(q) = (40 * q - 1184) / 9 82 <= q < 128
(16 * q + 128) / 9 64 <= q < 82

TVBR is not available for HE-AAC.

== Format support ==
Apple's CoreAudioBaseTypes lists the following AAC-derived codecs on macOS 11.3:<ref name=CABT>https://github.com/phracker/MacOSX-SDKs/blob/041600eda65c6a668f66cb7d56b7d1da3e8bcc93/MacOSX11.3.sdk/System/Library/Frameworks/CoreAudioTypes.framework/Versions/A/Headers/CoreAudioBaseTypes.h#LL391C2-L391C</ref>
* AAC-LC (MPEG-4)
* AAC-LD
* AAC-ELD
* AAC-ELD + SBR
* HE-AAC v1 (AAC + SBR)
* HE-AAC v2 (v1 + PS)
* USAC
* MPEG-4 Spatial

There should also be support for a number of other format, at least in decoding: Apple does not ship a standalone AAC codec, but a whole audio-processing kit.

Actual support depends on library version and whether the program used to call the library knows how to use these formats.
* afconvert is clumsy but exposes almost everything.
* QAAC only does LC and HEv1. It can use some of the audiotoolkit filters while it works.
* FFmpeg only knows about using Apple AAC with LC, LD, ELD, HE, and HEv2.

== afconvert ==
Apple's macOS includes a command-line utility for transcoding audio files that employs this encoder when encoding to AAC.

Usage:
afconvert [options] <infile> <outfile>

Options:
;-f --file <string>:File format. Several; use 'm4af' for '.m4a' file or 'adts' for raw '.aac' file.
;-d --data <string>:Data format. Several; use 'aac' for AAC-LC. See CoreAudioBaseTypes.h link in footnote.<ref name=CABT/>
;-s --strategy <0,1,2,3>:Bitrate allocation strategy. 0 for CBR, 1 for ABR, 2 for VBR_constrained, 3 for VBR
;-b --bitrate <n>:Bitrate in bits per second. Does nothing when used with strategy 3.
;-ue vbrq <0-127>:Output quality for VBR mode. Only applicable when using strategy 3.
;-q --quality <0-127>:Speed/quality trade-off. Internally rounded to a value of either 32, 64, or 96.

afconvert has way more options than those listed above, but not all are useful for AAC. Run <code>afconvert --help</code> to see them all, or see [https://ss64.com/osx/afconvert.html SS64] for an unofficial list.

== QAAC ==
{{Software Infobox
| name = qaac
| logo =
| screenshot =
| caption =
| maintainer = nu774
| stable_release = 2.80
| preview_release =
| operating_system = Windows
| use = Encoder
| license = Various
| website = [https://github.com/nu774/qaac github/qaac]
}}

QAAC is an open-source wrapper for ''Core Audio's'' AAC and [[ALAC]] encoders that allow them to be used directly in applications that don't otherwise use ''Core Audio''. It exists only for Windows.

* [https://github.com/nu774/qaac/wiki QAAC documentation]

Linux users [https://www.andrews-corner.org/qaac.html should be able to use wine]. The most fragile part apparently is [https://github.com/enzo1982/freac/issues/168 getting AppleApplicationSupport to install].

 

== ffmpeg ==
On macOS, ffmpeg 4.4 and above can use the <code>aac_at</code> codec to use Apple AAC via the new AudioToolbox.framework. In addition to AAC-LC, HE-AAC (v1/v2), AAC-LD, and AAC-ELD are available, the same selection of profiles as FDK AAC. FFmpeg uses a 0-14 quality range to be converted to the 0-127 scale, using the expression <code>q = 127 - q * 9;</code>.<ref>https://github.com/FFmpeg/FFmpeg/blob/f009f849617ab0c2015d2a748892d55b37d378c0/libavcodec/audiotoolboxenc.c#L328</ref>

On Windows, ffmpeg's build process can be tricked into linking to an CoreAudioToolbox.dll commonly distributed alongside iTunes, using [https://github.com/dantmnf/AudioToolboxWrapper AudioToolboxWrapper].

==External links==
<references />
*[http://www.apple.com/quicktime Apple's Quicktime Website]

Opus

2023-07-06T03:20:25Z

Logan589XP: Added release notes for libopus 1.4 and updated stable release number to 1.4.

{{Software Infobox
| name = Opus
| logo = [[Image:opus-logo.png|250px|Official Opus logo]]
| screenshot =
| caption = Opus Interactive Audio Codec
| maintainer = [http://xiph.org/ Xiph.Org Foundation]
| stable_release = 1.4
| operating_system = Windows, Mac OS/X, Linux/BSD
| use = Encoder/Decoder
| license = 3-clause BSD license
| website = [http://www.opus-codec.org/ opus-codec.org]
}}

'''Opus''' is a [[lossy]] audio compression format developed by the Internet Engineering Task Force (IETF) designed to be suitable for interactive real-time applications over the Internet,{{ref|homepage|a}} including music as well as speech, yet it is also very competitive for use as a storage and playback format, being a [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ class leader at around 64 kbps] and [http://listening-test.coresv.net/results.htm also at 96 kbps]. As an open format standardised through [http://tools.ietf.org/html/rfc6716 Request for Comments (RFC) 6716],{{ref|RFC|c}} a high quality reference implementation is provided under the 3-clause BSD license{{ref|homepage|a}} which compiles and runs on the vast majority of general purpose and embedded (fixed point) processors. Many Software patents which cover Opus are licensed under royalty-free terms.{{ref|FAQ|b}} Opus is also a Mandatory To Implement (MTI) codec for the upcoming WebRTC (Web Real Time Communication) specification of the World Wide Web Consortium (W3C).

Opus incorporates technology from two codecs, the speech-oriented SILK codec developed by Skype and the multi-purpose low-latency CELT codec developed by Xiph.org with significant changes to each to ensure they can work together.{{ref|RFC|c}} Opus can seamlessly transition among high and low bitrates, using a linear prediction codec (the SILK layer) at lower bitrates and a lapped transform codec (the CELT layer) at higher bitrates, as well as a hybrid of the two for a short overlap in which SILK encodes the 0–8 kHz spectrum and the CELT layer encodes only the frequencies above 8kHz.{{ref|RFC|c}} Opus has very low algorithmic delay (typ 22.5 ms) compared to popular music formats such as [[MP3]], [[Vorbis |Ogg Vorbis]], [[AAC | LC-AAC and HE-AAC]] (all over 100 ms), yet performs very competitively with them in terms of quality per bitrate, making it comparably viable as a storage & playback format. Also unlike Vorbis, Opus does not require the definition of large codebooks for each individual file, making it also preferable for short clips of audio, such as those often used by game developers, a field where patent-free Vorbis is commonly used.{{ref|RFC|c}}

Considerably more details of the history and potential applications for Opus are included in the ''Wikipedia'' page for '''[http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Opus (audio format)]'''

==Characteristics==
Opus supports bitrates from 6 kbps to 510 kbps for typical stereo audio sources (and a maximum of around 255 kbps per channel for multichannel audio), with the 'sweet spot' for music and general audio around 30 kbps (mono) and 40–100 kbps (stereo). It is intrinsically [[VBR | variable bitrate]], though constrained VBR and [[CBR | constant bitrate]] modes are possible where required. In the case of the reference release, libopus, the target bitrate is calibrated against the internal constant quality targets so that over a typical music collection, something very close to the target bitrate will be achieved. This bitrate-calibrated approach differs from most VBR encoders (e.g. LAME, helix mp3, qaac, Nero aacenc, Ogg Vorbis, Musepack) where a setting on some 'constant quality' scale (which differs between encoders) is used and the bitrate will fall where it may. Improved future versions can be expected to offer improved quality at the same setting. Independent implementations may adopt a different approach.

Opus is able to seamlessly adapt its mode of operation without glitches or sound interruption (an illustrative demonstration of [http://opus-codec.org/examples/#gauge bitrate scalability] is on the Opus Examples page), which can be particularly useful for mixed-content audio or varying network conditions, making the unified Opus codec superior to a suite of different codecs that might otherwise cover the same range of bitrate and quality settings and would require out-of-band signalling to instigate codec switching. The switching includes the choice of mono, stereo and other channel mappings, the use of the speech-oriented SILK layer, the general-purpose CELT layer or the hybrid of both, and the use of different audio bandwidths (4, 6, 8, 12, or 20 kHz) as well as the quality adjustments within the same operating mode that are available in most VBR-capable codecs.

Of importance mainly to interactive uses, but potentially useful in time-delayed audio streaming also, Opus includes packet loss concealment (PLC) in all modes and, in the speech-oriented modes where the SILK layer is active it also supports Forward Error Correction (FEC) where the expected rate of packet loss can be indicated to the encoder by the user or by application software and critical frames (e.g. consonant sounds) can be retransmitted at low bitrate to preserve intelligibility.

For music and general audio, the CELT layer of Opus builds on knowledge gained during xiph.org's Vorbis development and ensures as a primary goal that the total energy in each spectral band is preserved while requiring only a modest bitrate overhead to achieve this, thereby eliminating a lot of bitrate-starvation artifacts such as 'birdies' that are common in low-bitrate MP3, especially during transients, applause and cymbal sounds. This technique likewise increases coding efficiency at bitrates targetting transparent music reproduction. Short blocks (2.5 ms) are also possible for efficient transient handling. Short blocks can also be used exclusively, if very low algorithmic delay (5.0 ms) is required to enable very low-latency interative audio (e.g. live networked music performances such as remote jam sessions), though greater bitrate is then required to maintain the same quality (illustrated in [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo Monty's CELT demo page] under Constant PEAQ value, varying latency). CELT uses a number of additional techniques and provides additional advanced tools to enable encoder tuning.

Opus natively supports [[gapless playback]] (though [[Gapless_playback#Poorly_designed_playback_systems | poor player design]] might itself induce interruptions during playback). Playback gain is also required, making some form of [[ReplayGain]] or [[ReplayGain_2.0_specification | similar]] volume control possible in any compliant player.

==Bitrate performance==
For mono speech, Opus ranges from intelligible narrowband speech reproduction starting at 6 kbps to medium-band, wideband and superwideband speech, reaching full-band speech by around 14 kbps in encoder version 1.2 (was 21 kbps in v1.1, 29 kbps in v1.0). Above about 32 kbps, the SILK layer is no longer used at all, as CELT alone gives superior quality.

For music, the SILK modes are quite tolerable and better than CELT at very low bitrates. The hybrid mode is adopted as bitrate increases, extending bandwidth first to 12 kHz (comparable with compact cassette) then to the full 20 kHz and CELT then takes over. Assuming the source is stereo, the transition from mono to stereo typically happens between the transition from 12 kHz to 20 kHz. Encoder version 1.2 includes great improvements to music encoding in the 32–64 kbps range, allowing full-band stereo at 32 kbps and providing acceptable quality at 48 kbps where artifacts are audible but rarely annoying. Version 1.3 is expected to further improve quality in this range.

Multi-format stereo music listening tests have demonstrated the superiority of Opus at 64 kbps and 96 kbps compared to the best AAC-LC, HE-AAC and Ogg Vorbis encoders, and at 96 kbps also to 128 kbps MP3 encoded using LAME <code>-V 5</code>.

==Indicative bitrate and quality==
The tables below give illustrative, indicative quality guidance based on typical modes used internally by Opus and a range of listening tests.

In encoder version 1.1 automatic detection of speech/music and bandwidth detection were introduced to improve mode decisions and VBR is less constrained, all with the aim of maximizing the quality/bitrate tradeoff, and these improvements are further enhanced in version 1.2 and 1.3. These tables are likely to require updates as the encoder is improved, especially in low-bitrate regions.

===Speech encoding quality===
This table assumes a '''monophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate) but mentions stereo compatibility for 40kbps+. The default 20ms frame size (22.5ms latency) is assumed. Note that the selection of ''VOIP'' mode will deliberately modify the sound with a High Pass Filter and emphasis of formants and harmonics to improve intelligibility of speech especially in noisy environments much as telephones do. ''Auto'' mode will not modify the sound prior to encoding so is usually better for high quality speech recordings or mixed speech and music.

{| class="wikitable" style="text-align:center"
|-
!Bitrate Target
!Bandwidth
!Typical Mode Used
!Speech Quality
!Use Cases / Competitive Codecs
|-
!Less than 5 kbps
| —
| —
| Bitrates lower than 6 kbps not supported by Opus (SILK disabled if forced to encode, which results in terrible speech quality)
| Try [https://en.wikipedia.org/wiki/Codec_2 Codec 2] for 0.45–3.2 kbps mono speech or [[Wikipedia:Lyra (codec)|Lyra]] for 3.2 kbps mono speech
|-
!6 kbps
|6 kHz medium-band
|SILK
|Fair, intelligible
|AMR-NB may be a little better, but higher latency & proprietary, [[Speex]] also competitive
|-
!8 kbps
|6 kHz medium-band
|SILK
|Close to telephone quality
|AMR-NB & AMR-WB similar quality, but higher latency & proprietary. [[Speex]] competitive.
|-
!12 kbps
|12 kHz super-wideband
|hybrid
|Medium bandwidth, better than telephone quality
|Similar quality to AMR-WB
|-
!16 kbps
|20 kHz
|hybrid/CELT
|Wideband speech quality
|Similar to/better than AMR-WB
|-
!24 kbps
|20 kHz
|hybrid/CELT
|Near transparent speech
|Better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!32 kbps
|20 kHz
|CELT
|Essentially transparent speech plus moderately good stereo music
|Much better than AMR-WB. Podcasts/audiobooks/talk-radio.
|-
!40 kbps
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, fairly good stereo music
|Stereo podcasts/audiobooks/talk radio with some music
|-
!48 kbps or more
|20 kHz
|CELT
|Essentially transparent mono or stereo speech, reasonable music
|Flexible general purpose modes to suit mixed music and speech
|-
|}

===Music encoding quality===
This table assumes a '''stereophonic''' source sampled at CD quality or above (typ 48 kHz sampling rate). Opus will automatically use mono at very low bitrates, though a certain amount of stereo encoding can still be used (content dependent) even when mono is specified as the typical stereo mode in the table below.

{| class="wikitable" style="text-align:center"
|-
!Bitrate target
!Stereo mode
!Bandwidth
!typ SILK/CELT use
!Music quality notes
!Use cases/notes/competitive codecs
|-
!6 kbps
|mono
|6 kHz
|SILK
|Poor, muffled sound but intelligible lyrics.
| —
|-
!8 kbps
|mono
|6 kHz
|SILK
|Poor, muffled but OK for bitrate
| —
|-
!14 to 16 kbps
|mono
|20 kHz
|hybrid/CELT
|Fairly poor but OK for bitrate
|Perhaps acceptable for incidental music
|-
!22 to 24 kbps
|mono
|20 kHz
|hybrid/CELT
|Fair but OK for bitrate
|OK for incidental music
|-
!32 to 40 kbps
|stereo
|20 kHz
|CELT
|Moderately good stereo, some artifacts, rarely nasty
|Stereo podcasts, audiobooks, very low bitrate music
|-
!48 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, may have problems with cymbals
|Stereo podcasts, audiobooks, low bitrate music
|-
!64 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, nice sound, detectable differences to original (mostly 'not annoying')
|Music storage & streaming. Beat HE-AAC, Vorbis, MP3 in [https://web.archive.org/web/20200130083553/http://people.xiph.org/~greg/opus/ha2011/ listening test]
|-
!96 kbps
|stereo
|20 kHz
|CELT
|Full bandwidth stereo music, good quality approaching transparency
|Music storage & high quality streaming. Beat LC-AAC, Vorbis, MP3 in [http://listening-test.coresv.net/results.htm listening test]
|-
!112 kbps
|stereo
|20 kHz
|CELT
|Fairly close to transparency (needs more testing)
|Music storage & high quality streaming. Very low-latency stereo networked music performance/jam sessions at OK quality (see below table)
|-
!128 kbps
|stereo
|20 kHz
|CELT
|Very close to transparency (needs more testing). Most modern codecs competitive (AAC-LC, Vorbis, MP3)
|Music storage & streaming. Future download music sales.
|-
!160 to 192 kbps
|stereo
|20 kHz
|CELT
|Transparent with very low chance of artifacts (a few killer samples still detectable). Most old & new lossy codecs competitive.
|Music storage & streaming, dedicated limited-bandwidth audio links (e.g. wireless, [http://en.wikipedia.org/wiki/Bluetooth_profile#Advanced_Audio_Distribution_Profile_.28A2DP.29 A2DP-bluetooth] type links).
|-
!510 kbps
|stereo
|20 kHz
|CELT
|Maximum possible stereo bitrate target (actual rate often less than 510 for default frame size). Most old and new lossy codecs competitive, plus near-lossless [[lossyWAV]] and [[WavPack | WavPack lossy]]
|Music storage, dedicated limited-bitrate audio links (e.g. wireless, minimum latency high quality audio. LossyWAV and WavPack lossy are very competitive for storage, and WavPack lossy <code>--blocksize=256</code> may be competitive with minimum latency mode also.
|-
!>510 kbps
| —
| —
| —
|Above Opus bitrate range allowed for stereo sources
|Settle for 510 kbps or use [[lossless]], [[lossyWAV]], [[WavPack | WavPack lossy]] or lossy transform/subband codecs like [[Vorbis]], [[Musepack]] at very high settings.
|-
|}

===Lower latency versus quality/bitrate trade-off===
====Packet overhead in interactive applications====
For interactive use on the Internet or other packet-based networks, total bandwidth used will be subject to packet overhead. The more packet headers that are transmitted every second, the greater will be the overhead that is required. For this reason, Opus, while defaulting to 20 ms frames, supports 60 ms frames to reduce overhead when transporting low-bitrate SILK frames at the expense of greater latency, which may still be acceptable for speech, and also supports 10 ms SILK frames to reduce latency somewhat at the expense of packet overhead.

In the CELT layer, which tends to operate at higher bitrates than SILK, 20 ms frames are the default, but frames of 10 ms, 5 ms and 2.5 ms are also possible, which directly increases the frame overhead by transmitting more packets per second to achieve lower latency. In addition, as we'll see below it also reduces the quality/bitrate tradeoff of the CELT layer itself.

You probably do not want to use a frame size lower than 10 ms in applications containing speech, as doing so turns off SILK. The "lowdelay" application switch (available in FFmpeg and the raw library) turns off SILK to cut out 4 ms of synchronization delay, but a frame size of 10 ms achieves more delay reduction compared to default without sacrificing SILK.

None of the bitrates mentioned in this article account for the packet overhead.

====CELT layer latency versus quality/bitrate trade-off====
Unlike the SILK layer, which works on fixed 10 ms blocks, 1, 2 or 6 of which can be combined into an Opus frame, the CELT layer is able to modify the encoding block lengths available to enable its use with shorter frames.

When the CELT layer uses 10 ms, 5 ms and 2.5 ms frames instead of the default 20 ms, it must use smaller transform block sizes to achieve this, thereby reducing frequency resolution in the MDCT compared to the default transform window, thus reducing encoding efficiency for tonal signals. To obtain the same frequency precision for a sound divided into shorter transform windows, improved amplitude precision is necessary, resulting in increased bitrate to obtain the same perceptual quality (or conversely lower quality at the same bitrate).

These reduced-latency modes remain efficient for transient signals, which use short blocks anyway.

In all modes, the algorithmic delay consists of the frame size plus an additional 2.5 ms delay. The CELT layer requires 2.5 ms for MDCT window overlap.

Xiph.org used matched PEAQ scores (approximate perceptual quality assessment made in software) for the CELT0.10 codec that was used as the basis of the CELT layer in the Opus reference release, which indicate the following [http://people.xiph.org/~xiphmont/demo/celt/demo.html#demo approximate equivalent settings] for stereo music.

{| class="wikitable" style="text-align:center"
|-
!Frame size
!Algorithmic delay
!Bitrate to match 64kbps@22.5ms delay
!fractional bitrate increase
|-
!20 ms
|22.5 ms
|64.0 kbps
| +0.0 %
|-
!10 ms
|12.5 ms
|70.4 kbps
| +10.0 %
|-
!5 ms
|7.5 ms
|84.8 kbps
| +32.5 %
|-
!2.5 ms
|5.0 ms
|112.0 kbps
| +75.0 %
|-
|}

N.B. This table is useful for interactive streaming only. For music storage & delayed playback or non-interactive streaming, latency reduction is not important and the default 20 ms frame size is preferable.

== Implementations ==

The format and algorithms are openly documented and the reference implementation is published as free software. The reference implementation (Opus Audio Tools, opus-tools), consisting of separate encoders and decoders, is published under the terms of a BSD-like license. It is written in C programming language and can be compiled for hardware architectures with or without floating point unit. The accompanying diagnostic tool opusinfo reports detailed technical information about Opus files, including information on the standard compliance of the bitstream format. It is based on ogginfo from the vorbis-tools and therefore, unlike the encoder and decoder, available under the terms of version 2 of the GPL.

=== Reference implementation (libopus + binaries) ===
The commandline tools of the reference version are available pre-compiled for the most popular operating systems at [http://opus-codec.org/downloads opus-codec.org] and [https://ftp.mozilla.org/pub/mozilla.org/opus/ Mozilla's ftp server], plus in the foobar2000 free encoders pack and some alternative compiles through the hydrogenaud.io opus forum. The libopus commandline tools include encoder <code>opusenc</code>, decoder <code>opusdec</code>, and with a different license, the <code>opusinfo</code> opus stream & metadata analyzer.

The '''latest stable release''' is recommended for general use and as of mid 2014 is considered competitive with or superior to the best alternative speech or general music encoders at most supported bitrates.

==== libopus v1.0 ====
Released 11 Sep 2012 when RFC6716 was standardized but mostly fully developed by late 2011.

'''Stable, well-tuned''' <code>opusenc</code> reference encoder as included in RFC documentation.

CELT layer closely related to CELT 0.10 implements Constrained VBR mode by default (bitrate boost used mainly for transients), plus true CBR.

==== libopus v1.1 ====

The alpha source code released 21 Dec 2012 for testing & user feedback and following a beta release and testing, the stable 1.1 version was released on 5 December 2013, considered well tested enough for general release.

CELT layer [http://jmspeex.livejournal.com/11737.html quality improvements] introduced to provide '''unconstrained VBR''' include a rate boost not just for transients but now for highly tonal signals too and rate reduction when stereo image is narrow. There's also a rewrite of its '''transient detection''' code and '''time-frequency analysis''' code, and rewritten '''dynamic allocation''' code (HF/LF tilt and Band Boost) to allow more aggressive changes from the typical static allocation when warranted.

There are many minor improvements to '''speech quality''' in both SILK and CELT layers.

*'''DC-rejection''' below 3 Hz also aids quality if inaudible DC offset is present with no effect on deep bass notes.
*'''Automatic speech/music detection''' is introduced to optimize encoding mode choices, especially near the bitrate target range (presumably around 24–40 kbps) where the encoder may perform best with SILK, hybrid or CELT depending on content type. Below that range SILK performs best for both music & speech, and above it CELT performs best for speech & music. The detection, without look-ahead is not perfect but usually is undecided in audio where either mode will work well.
*'''Automatic bandwidth detection''' is also introduced to save wasted bits allocated to absent frequencies.
*'''Surround sound improvements''' were introduced since the beta release with considerable advances in coding efficiency, bitrate allocation and quality.

==== libopus v1.1.3 ====
Released July 15th, 2016. This version contains:

*Neon optimizations improving performance on ARMv7 and ARMv8 by up to 15%
*Fixes some issues with 16-bit platforms (e.g. TI C55x)
*Fixes to comfort noise generation (CNG)
*Documenting that PLC packets can also be 2 bytes
*Includes experimental ambisonics work (<code>--enable-ambisonics</code>)

==== libopus v1.2.1 ====
Released June 26th, 2017. This version contains:

*Speech quality improvements especially in the 12–20 kbit/s range
*Improved VBR encoding for hybrid mode
*More aggressive use of wider speech bandwidth, including fullband speech starting at 14 kbit/s
*Music quality improvements in the 32–48 kbit/s range
*Generic and SSE CELT optimizations
*Support for directly encoding packets up to 120 ms
*DTX support for CELT mode
*SILK CBR improvements
*Support for all of the fixes in draft-ietf-codec-opus-update-06 (the mono downmix and the folding fixes need <code>--enable-update-draft</code>)
*Many bug fixes, including integer wrap-arounds discovered through fuzzing (no security implications)

==== libopus v1.3 ====
Released on October 18th, 2018. This version contains:

* Improvements to voice activity detection (VAD) and speech/music classification using a recurrent neural network (RNN)
* Support for ambisonics coding using channel mapping families 2 and 3
* Improvements to stereo speech coding at low bitrate
* Using wideband encoding down to 9 kb/s
* Making it possible to use SILK down to bitrates around 5 kb/s
* Minor quality improvement on tones
* Enabling the spec fixes in <nowiki>RFC 8251</nowiki> by default
* Security/hardening improvements
* Fixes to the CELT PLC
* Bandwidth detection fixes

==== libopus v1.3.1 ====
Released on April 12th, 2019. This version contains:

* Fixes to x87 builds
* A new OPUS_GET_IN_DTX query to know if the encoder is in DTX mode (last frame was either a comfort noise frame or not encoded at all)
* A new (and still experimental) CMake-based build system that is eventually meant to replace the VS2015 build system (the autotools one will stay)

==== libopus v1.4 ====
Released on April 20th, 2023. This version contains:

* Improved tuning of the Opus in-band FEC (LBRR). See the issue for details
* Added a OPUS_SET_INBAND_FEC(2) option that turns on FEC, but does not force SILK mode (FEC will be disabled in CELT mode)
* Improved tuning and various fixes to DTX
* Added Meson support, improved CMake support In addition to the improvements above, this release includes many minor bug fixes.

=== Other implementations ===

==== Concentus ====

The libopus reference library (fixed-point variant) has successfully been ported to both '''C#''' and '''Java''', as part of a project called '''Concentus'''. The aim of the project is specifically to target cross-platform applications where native C interop is relatively difficult. The code is available on [https://github.com/lostromb/concentus Github] and distributed via standard package managers.

==== Emscripten ports ====

At least one port of reference opus in Javascript has been made using the automated tool [https://developer.mozilla.org/en-US/docs/Mozilla/Projects/Emscripten emscripten]. See [https://blog.rillke.com/opusenc.js/ here], [https://github.com/kazuki/opus.js-sample here] and [https://github.com/audiocogs/opus.js here].

==== ffmpeg ====
FFmpeg has a native [https://ffmpeg.org/ffmpeg-codecs.html#opus "opus"] codec. It is of lower quality than the reference libopus and only does CELT coding. However, it is still good for the ecosystem to have a completely independent implementation.

== Hardware & Software Support ==

Much of this section is based heavily on the Jan 12th 2013 version of the '''Support''' section of the [http://en.wikipedia.org/wiki/Opus_%28audio_format%29 Wikipedia article], which is more likely to be kept updated and to provide links to further information about the supporting platforms.

=== VoIP software ===
* The open source virtual PBX Freeswitch supports Opus transcoding.
* The voice-chat software Mumble supports Opus as its main codec.
* SIP softphones Phoner and PhonerLite support Opus
* The SIP and IAX2 client SFLphone is being fitted with Opus support.
* Integration of Opus into the Skype client is finished, although no version with Opus support has yet been published.
* TrueConf video conferencing solutions support Opus.
* Opus support is planned for Jitsi 2.0, together with VP8 video.
* Empathy may use any format supported in GStreamer, including Opus.
* Line2 has replaced their current codec with Opus. Their iOS app will be the first to be released with the Opus. The Android app will follow later.
* CSipSimple supports Opus, Codec2, G.726 and G.722.1 with an additional plug-in.
* The voice-chat software TeamSpeak 3 supports Opus for voice and music in pre-release server 3.0.7-pre2 and beta client version 3.0.10.
* The proprietary instant messenger service Discord uses Opus audio for all voice calls and video calls, regardless of platform.

=== Web frameworks and browsers ===
* Opus support is mandatory for WebRTC implementations.
* Mozilla supports Opus beginning with version 15 of Firefox and Thunderbird, plus Seamonkey, which uses a shared codebase.
* Depending on the backend in use, Opera supports inline playback of embedded Opus files. Official support for Opus and WebRTC are on the development roadmap.
* Chromium and Google Chrome have audio support as of version 33.
* Apple's Safari browser now supports Opus as of iOS 11 and macOS 10.13 High Sierra.
* Maxthon Cloud Browser

=== Streaming audio ===
* Icecast. (examples: [http://dir.xiph.org/by_format/Opus Stream directory by format Opus], [http://smj.delfa.net/opus_64.m3u 64k]/[http://smj.delfa.net/opus_256.m3u 256k] [http://smj.delfa.net/ Smooth Jazz Opus Stream], [http://www.absoluteradio.co.uk/listen/labs.html Absolute Radio Opus Trial] 7 stations at 24,64,96 kbps, [http://icecast.ofdoom.com:8000/burst-opus.ogg Icecast Of Doom 96k]
* Krad Radio
* Liquidsoap

=== Operating systems and desktop multimedia frameworks ===
* In Debian GNU/Linux the Opus development tools and supporting libraries can be installed from the preconfigured repositories in the next stable version ("wheezy") that is expected to be released in early 2013.
* For Microsoft Windows, there are DirectShow filters supporting Opus, including DC-Bass Source Mod and the LAV Filters.
* In GStreamer the integration of Opus support is complete.
* FFmpeg supports decoding and encoding Opus via the external library libopus.
* Android 5.0 and above supports Opus natively if encapsulated in the Ogg container, but .opus filename extension is not recognized by Android, so the use of double filename extension .opus.ogg is recommended as a workaround to allow apps to recognize files as playable audio.

=== Hardware support ===
* Support in [[Rockbox]] is available. This means hardware support for a series of portable media players (including some products from the iPod series by Apple and Sansa, iriver and Archos devices) and with "Rockbox as an Application" (RaaA) also on Android devices.

=== Player software ===

* Windows/Mac/Linux (Cross-Platform)
*# [[VLC]] (media player supports Opus as of version 2.0.4
*# [[Amarok]] 2.8 has transcoding support for Opus codec if ffmpeg is compiled with support for the libopus library & support for playback of Opus encoded files if Amarok is compiled against TagLib (newer than V1.8)
*# Clementine has Opus support
*# Audacious player
*# [[MPD]] as of version 0.18 if compiled against libopus (supports both encoding for http streams and decoding)
* Windows Exclusive
*# AIMP supports Opus natively as of version 3.20 build 1125 beta 1
*# [[foobar2000]] supports Opus natively as of v1.1.14 beta 1
*# Mpxplay supports Opus (using a decoder DLL) as of v1.60 alpha 2
*# [[Winamp]] supports Opus using a [http://forums.winamp.com/showthread.php?p=2925154#post2925154 3rd party plug-in]
*# MPC-HC
*# Resonic Player/Pro supports Opus natively as of version 0.2.2
* iOS/Android (Cross-Platform)
*# Capriccio [https://itunes.apple.com/us/app/capriccio-free-ultimate-music/id434829018?mt=8 iOS]/[https://play.google.com/store/apps/details?id=me.ideariboso.capriccio Android]
*# foobar2000 [https://itunes.apple.com/us/app/foobar2000/id1072807669?mt=8 iOS]/[https://play.google.com/store/apps/details?id=com.foobar2000.foobar2000&hl=en Android]
* Android Exclusive
*# [https://play.google.com/store/apps/details?id=in.krosbits.musicolet Musicolet Music Player]
*# [http://gonemadmusicplayer.blogspot.com/ GoneMAD Music Player]
*# [http://neutronmp.com/ Neutron Music Player]
*# [http://www.videolan.org/vlc/download-android.html VLC Media Player for Android]
*# [https://play.google.com/store/apps/details?id=ru.recoilme.freeamp FreeMP]
*# [https://play.google.com/store/apps/details?id=net.mderezynski.youki3 Youki]
*# [https://play.google.com/store/apps/details?id=com.aimp.player AIMP for Android]
*# [https://play.google.com/store/apps/details?id=com.acmeandroid.listen Listen Audiobook Player]
*# [https://play.google.com/store/apps/details?id=com.mxtech.videoplayer.ad MX Player]
*# [https://play.google.com/store/apps/details?id=org.tomahawk.tomahawk_android Tomahawk Player Beta]
*# [https://play.google.com/store/apps/details?id=com.maxmpz.audioplayer&hl=en Poweramp Music Player]

=== Other software ===
* CDBurnerXP
* MediaCoder
* Report-IT
* [[MP3tag|MP3tag]]
* [https://moisescardona.me/opus-gui/ Opus GUI]
* [http://www.xdlab.ru/en/ TagScanner]
* [http://www.xmedia-recode.de/ XMedia Recode]

== References & Notes ==

*{{note|homepage|a}}[http://opus-codec.org/ opus-codec.org homepage]
*{{note|FAQ|b}}[http://wiki.xiph.org/OpusFAQ Opus FAQ]
*{{note|RFC|c}}[http://tools.ietf.org/html/rfc6716 IETF RFC 6716]

[[Category:Codecs]]
[[Category:Lossy]]
[[Category:Encoder/Decoder]]

Foobar2000:Free Encoder Pack

2023-07-06T03:00:02Z

Logan589XP: Updated release date and encoder versions to reflect the newest Encoder Pack release from July 5th, 2023.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2023-07-05
| stable_release_date = 2023-07-05
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.80
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.80
|
|-
| [[FLAC]]
| flac
| 1.4.3
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.4
|-
| [[WavPack]]
| wavpack
| 5.6.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Foobar2000:Free Encoder Pack

2023-07-04T20:31:19Z

Logan589XP: Updated release date and encoder versions to reflect the newest Encoder Pack release from July 4th, 2023.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2023-07-04
| stable_release_date = 2023-07-04
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.79
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.79
|
|-
| [[FLAC]]
| flac
| 1.4.3
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.4
|-
| [[WavPack]]
| wavpack
| 5.6.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Foobar2000:Free Encoder Pack

2023-05-31T01:15:52Z

Logan589XP: Updated release date and encoder versions to reflect the newest Encoder Pack release from April 30th, 2023.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2023-04-30
| stable_release_date = 2023-04-30
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.79
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.79
|
|-
| [[FLAC]]
| flac
| 1.4.2
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.4
|-
| [[WavPack]]
| wavpack
| 5.6.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]

Nero AAC

2023-05-31T00:48:17Z

Logan589XP: Fixed "neroAacEnc" final version (1.5.4.0) release year from 2018 to 2010.

[[category:Encoder/Decoder]]
{{Software Infobox
| name = Nero AAC Codec
| logo =
| screenshot =
| caption =
| maintainer = Nero AG
| stable_release = [[#Versions|See table]]
| preview_release = n/a
| operating_system = Windows, Linux
| use = Encoder/Decoder
| license = see site for license
| website = [https://web.archive.org/web/20160304031341/http://www.nero.com/enu/company/about-nero/nero-aac-codec.php Information] [https://web.archive.org/web/20160923100008/http://ftp6.nero.com/tools/NeroAACCodec-1.5.1.zip NeroAACCodec-1.5.1.zip]
}}
{{aac-encoders}}
'''Nero AAC''' is a freely available software suite that includes a high-quality LC and HE [[AAC]] [[codec]] and metadata tool. It was released by Nero AG for personal non-commercial and/or technology-evaluation purposes. The download package contained binaries for both Windows and Linux. It was the same codec that was used in Nero's other, then current, commercial products.

It was generally perceived to have the highest quality [[VBR]] HE-AAC and good enough LC-AAC implementation in past. The codec could also create HEv1/v2 AAC streams for extremely low bitrates and supports multi-channel surround sound encoding.

== Versions ==
The suite includes three command-line tools.

{| class="wikitable"
! Tool !! Role !! Latest Version
|-
| [[#neroAacEnc|neroAacEnc]] || AAC encoder || 1.5.4.0 (Feb 18 2010)
|-
| [[#neroAacDec|neroAacDec]] || AAC decoder || 1.5.1.0 (Dec 17 2009)
|-
| [[#neroAacTag|neroAacTag]] || MP4 metadata editor || 1.5.1.0 (Dec 17 2009)
|}

=== History ===
The codec was originally part of Nero Digital, a complete [[MPEG-4]] Audio/Video solution. The ASP/AVC (video) codec was developed by a French company called Ateme. Nero built an in-house team to develop the AAC (audio) codec that included Ivan Dimkovic, Menno Bakker, and others. Dimkovic was the author of the PsyTel codec, and the Nero AAC codec is said to be based on this work. Menno Bakker was the developer of [[FAAC]], one of the earliest widely-available AAC encoders, and also what would be become its companion decoder, [[FAAC|FAAD]]. The Nero AAC codec became a stand-alone package around 2006, although still called Nero Digital Audio until 2009. Nero apparently still uses the codec in its products. Neither Dimkovic nor Bakker currently work at Nero, and development of the codec has stalled, but the software is stable and remains a reliable, and high-quality, option for AAC encoding. 

* Original release, version 1.0.0.0, on 2006-05-01.
* Tagging utility added in version 1.0.0.2 (2006-05-18).
* Linux officially supported in version 1.1.34.2 (2007-08-06).
* Renamed from Nero Digital Audio to Nero AAC in version 1.5.1.0 (2009-12-17).
* SSE2 no longer required in version 1.5.3.0 (2009-12-29).
* Disappeared from nero.com on 2014-07-23.
* Re-appeared on nero.com in 2014-09.

== NeroAacEnc ==
{{infobox software
| name = neroAacEnc
| logo =
| screenshot =
| caption =
| developer = Nero AG
| released = {{start date and age|2006|05|01}}
| stable_release = 1.5.4.0
| stable_release_date = {{start date and age|2010|02|18}}
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows, Linux
| use = Encoder
| license = Freeware
| website =
}}
This is the encoder command-line tool. It is used to encode a [[RIFF_WAVE|.WAV]] file containing PCM data into an [[MP4|.MP4]] file containing an AAC stream.

=== Usage ===

Windows
<pre style="display: table;">neroAacEnc.exe [options] -if <input-file> -of <output-file></pre>

Linux:
<pre style="display: table;">./neroAacEnc [options] -if <input-file> -of <output-file></pre>
Note: Make sure the binary is executable.

Where:
; <tt><input-file></tt>
:Path to source file to encode. The file must be in Microsoft WAV format and contain PCM data. Specify - to encode from stdin.
: Note that multiple input files can be specified, they will be encoded together into a single output file with chapter marks indicating source file divisions.
; <tt><output-file></tt>
: Path to output file to encode to, in MP4 format.

==== Options ====
===== Quality/bitrate control =====
; <tt>-q <number></tt>
: Enables "target quality" mode. (VBR) <tt><number></tt> is a floating-point number in 0...1 range. I.e. 0.5 or 0.75, etc. with 1 being the highest quality and thus also having the largest output file size.
; <tt>-br <number></tt>
: Specifies "target bitrate" mode. (ABR) <tt><number></tt> is target bitrate in bits per second ranging from 8000 to 400000.
; <tt>-cbr <number></tt>
: Specifies "target bitrate (streaming)" mode. (CBR) <tt><number></tt> is target bitrate in bits per second ranging from 8000 to 400000.

When neither of above quality/bitrate options is used, the encoder defaults to equivalent of <tt>-q 0.5</tt>

The following VBR quality settings result in the following average bitrates:
{| class="wikitable"
! Quality
! Bitrate (kbit/s), stereo
|-
| 0.05
| 16
|-
| 0.15
| 33
|-
| 0.25
| 66
|-
| 0.35
| 100
|-
| 0.45
| 146
|-
| 0.55
| 192
|-
| 0.65
| 238
|-
| 0.75
| 285
|-
| 0.85
| 332
|-
| 0.95
| 381
|}

===== Multipass encoding =====
; <tt>-2pass</tt>
: Enables two-pass encoding mode.
: Note that two-pass mode requires a physical file as input, rather than stdin.
: Using two-pass encoding is not recommended when using VBR.
; <tt>-2passperiod <number></tt>
: Overrides two-pass encoding bitrate averaging period, in milliseconds.
: Specify zero to use least restrictive value possible (default).

===== Advanced features / troubleshooting =====
; <tt>-lc</tt>
: Forces use of LC AAC profile (HE features disabled). (LC = Low Complexity)
; <tt>-he</tt>
: Forces use of HE AAC profile (HEv2 features disabled). (HE = High Efficiency)
; <tt>-hev2</tt>
: Forces use of HEv2 AAC profile

{{panel|'''Note:''' The above switches ({{code|-lc}}, {{code|-he}}, {{code|-hev2}}) are '''not required''' and should normally not be used. The optimal AAC profile is automatically determined from quality/bitrate settings when no override is specified. A scenario where using the switches might be useful is to force LC when you have a device that doesn’t support HE.|color=yellow}}

; <tt>-ignorelength</tt>
: Ignores length signaled by WAV headers of input file. Useful for certain frontends using stdin.

== NeroAacDec ==

NeroAacDec is the command-line decoder. It is currently at version 1.5.1.0 and was last updated Dec 17th, 2009. It decodes an [[MP4|.MP4]] file into a [[RIFF_WAVE|.WAV]] file

=== Usage ===

Windows:
<pre>neroAacDec.exe -if <input file> -of <output file> [-chapter <number>]</pre>

Where:
; <tt><input-file></tt>
: Path of source input file to decode, in MP4 format.
; <tt><output-file></tt>
: Path to decode the file to, in WAV format. Specify - to decode to stdout.

==== Options ====

; <tt>-chapter <number></tt>
: Specify the chapter mark within the input MP4 file.

== NeroAacTag ==

NeroAacTag is used to list or edit an MP4 file's meta-data. It is currently at version 1.5.1.0 and was last updated Dec 17th, 2009. By default is writes iTunes-compatible tags.

=== Usage ===

Windows:
<pre>neroAacTag.exe <file.mp4> <command> [<command> [<command> ...]]</pre>

==== Available commands ====
; <tt>-list-meta</tt>
: Lists existing metadata entries.
; <tt>-meta:<name>=<value></tt>
: Sets specified metadata field to specified value. Eg. -meta:artist="Pink Floyd"
; <tt>-meta-user:<name>=<value></tt>
: Sets specified metadata field to specified value. Allows non-standard metadata fields to be added. '''WARNING''': fields added using -meta-user are not guaranteed to be read back on all Nero Digital compliant software/hardware.
; <tt>-list-standard-meta</tt>
: Displays a list of field names usable with -meta command.
; <tt>-list-standard-itunes-meta</tt>
: Displays a list of iTunes field names. (Usable with -meta-user)
; <tt>-list-standard-memorystick-meta</tt>
: Displays a list of iTunes field names.
; <tt>-list-covers</tt>
: Lists cover art entries.
; <tt>-write-nd-covers</tt>
: Write cover art also to ND tags. Default is to write only to iTunes tags.
; <tt>-add-cover:<type>:<jpegfile></tt>
: Creates a cover art entry from specified JPEG file. <tt><type></tt> specifies type of cover art entry and can be "front" or "back". If specified cover art entry already exists, its contents are overwritten. Eg. <tt>-add-cover:back:hello.jpg</tt>
; <tt>-dump-cover:<type>:<jpegfile></tt>
: Dumps specified cover art entry contents to a JPEG file.
; <tt>-remove-cover:<type></tt>
: Removes specified cover art entry.
; <tt>-remove-cover:all</tt>
: Removes all cover art entries.
; <tt>-list-chapters</tt>
: Lists chapters present in the file.
; <tt>-chapter:<number></tt>
: Sets chapter index metadata edits apply to. Value of 0 (default state) applies edits to all present chapters. Also affects <tt>-list-meta</tt> output.
; <tt>-chapters-to-tracknumbers</tt>
: Generates track number metadata according to the chapter list.

=== Supported metadata fields ===

Strings are all encoded in utf8.

{| class="wikitable sortable"
! Tag name !!class="unsortable"| Usage !! MP4 block modified !!class="unsortable"| Comment
|-
| title || -meta:title=<string> || ©nam
|-
| artist || -meta:artist=<string> || ©ART
|-
| year || -meta:year=<string> || ©day || Block actually supports full YYYY-MM-DD
|-
| album || -meta:album=<string> || ©alb
|-
| genre || -meta:genre=<string> || gnre or ©gen || Stores an ID3 genre number (-1) as integer in the '''gnre''' block if one exists for the given string, or stores the string in the "user-defined" '''©gen''' block if not
|-
| track || -meta:track=<int> || trkn || Block stores both track and totaltracks in one binary value
|-
| totaltracks || -meta:totaltracks=<int> || trkn || Block stores both track and totaltracks in one binary value
|-
| disc || -meta:disc=<int> || disk || Block stores both disc and totaldiscs in one binary value
|-
| totaldiscs || -meta:totaldiscs=<int> || disk || Block stores both disc and totaldiscs in one binary value
|-
| url || -meta:url=<string> || ---- || Stored in com.apple.iTunes:url
|-
| copyright || -meta:copyright=<string> || ---- || Stored in com.apple.iTunes:copyright
|-
| comment || -meta:comment=<string> || ©cmt
|-
| lyrics || -meta:lyrics=<string> || ©lyr
|-
| credits || -meta:credits=<string> || ---- || Stored in com.apple.iTunes:credits
|-
| rating || -meta:rating=<?> || ---- || Stored in com.apple.iTunes:rating
|-
| label || -meta:label=<string> || ---- || Stored in com.apple.iTunes:label
|-
| composer || -meta:composer=<string> || ---- || Stored in com.apple.iTunes:composer (See writer for '''©wrt''')
|-
| isrc || -meta:isrc=<string> || ---- || Stored in com.apple.iTunes:isrc ([http://isrc.ifpi.org/en/ International Standard Recording Code])
|-
| mood || -meta:mood=<string> || ---- || Stored in com.apple.iTunes:mood
|-
| tempo || -meta:tempo=<int> || tmpo || Beats per minute, stored as a 16-bit integer
|-
| writer || -meta-user:writer=<string> || ©wrt
|-
| contentgroup || -meta-user:contentgroup=<string> || ©grp
|-
| album artist || -meta-user:'album artist'=<string> || aART
|-
| description || -meta-user:description=<string> || desc
|-
| network || -meta-user:network=<string> || tvnn
|-
| show || -meta-user:show=<string> || tvsh
|-
| season || -meta-user:season=<int> || tvsn || Stored as 32-bit integer
|-
| episode || -meta-user:episode=<int> || tves || Stored as 32-bit integer
|-
| episodename || -meta-user:episodename=<string> || tven
|-
| sorttitle || -meta-user:sorttitle=<string> || sonm
|-
| sortalbum || -meta-user:sortalbum=<string> || soal
|-
| sortartist || -meta-user:sortartist=<string> || soar
|-
| sortband || -meta-user:sortband=<string> || soaa
|-
| sortwriter || -meta-user:sortwriter=<string> || soco
|-
| sortshow || -meta-user:sortshow=<string> || sosn
|-
| itunescompilation || -meta-user:itunescompilation=<0,1> || cpil || iTunes only. More common to use something like "Various Artists" in the Album Artist field
|-
| itunespodcast || -meta-user:itunespodcast=<0,1> || pcst
|-
| tool || ''read-only'' || ©too || NeroAacEnc sets this value to "Nero AAC codec / <''version''>"
|-
| ''covers''
| -add-cover:<type>:<jpegfile> -dump-cover:<type>:<jpegfile> -remove-cover:<type> -remove-cover:all
| covr
| Types: front, back
|}

=== Notes ===

neroAacTag has a quirk related to its memory-stick tag capability. Meta tag values that contain one or more semi-colons (;) will get split up into multiples of that tag.

For example, you run the following command:

<pre>
neroAacTag test.m4a -meta:artist="Artist A; Artist B; Artist C"
</pre>

Instead of having a single artist tag listing all three artists (<tt>artist="Artist A; Artist B; Artist C"</tt>), you get three artist tags with an artist listed in each (<tt>artist="Artist A" artist=" Artist B" artist=" Artist C"</tt>).

neroAacTag will show your tag as correctly applied but iTunes or Mp3tag will not see that. (Mp3tag will show you the duplicate tags in the Extended tags view.)

== Examples ==

=== Linux ===
bash script to transcode FLAC to AAC in Linux preserving metadata:

<pre><nowiki>
#!/bin/bash

# Transcode FLAC to AAC (M4A) using:
# flac, metaflac, neroAacEnc, neroAacTag

# Creates M4A versions of all FLAC in the current directory.

AACENC="neroAacEnc"
AACTAG="neroAacTag"
TMPCVR="/run/shm/flac_cover.jpg"

gettagvalue_metaflac () {
local SRCF="$1"
local BLOCK="$2"
local VALUE=$(metaflac "$SRCF" --show-tag="$BLOCK")

shopt -s nocasematch
if [[ $VALUE =~ ^"$BLOCK"=(.*?) ]]; then
VALUE=${BASH_REMATCH[1]}
else
VALUE=""
fi
shopt -u nocasematch
echo "$VALUE"
}

for f in *.flac
do
TITLE=$(gettagvalue_metaflac "$f" "TITLE")
ARTIST=$(gettagvalue_metaflac "$f" "ARTIST")
ALBUMARTIST=$(gettagvalue_metaflac "$f" "ALBUMARTIST")
ALBUM=$(gettagvalue_metaflac "$f" "ALBUM")
TRACKNUMBER=$(gettagvalue_metaflac "$f" "TRACKNUMBER")
TRACKTOTAL=$(gettagvalue_metaflac "$f" "TRACKTOTAL")
DISCNUMBER=$(gettagvalue_metaflac "$f" "DISCNUMBER")
DISCTOTAL=$(gettagvalue_metaflac "$f" "DISCTOTAL")
DATE=$(gettagvalue_metaflac "$f" "DATE")
GENRE=$(gettagvalue_metaflac "$f" "GENRE")

if [ -e "$TMPCVR" ]; then
rm "$TMPCVR"
fi
metaflac --export-picture-to="$TMPCVR" "$f"

fm4a=$(echo "$f" | sed s/\.flac$/.m4a/g)
flac -s -d -c "$f" | "$AACENC" -ignorelength -q 0.5 -if - -of "$fm4a"

"$AACTAG" "$fm4a" \
-meta:title="$TITLE" \
-meta:artist="$ARTIST" \
-meta-user:'album artist'="$ALBUMARTIST" \
-meta:album="$ALBUM" \
-meta:year="$DATE" \
-meta:track="$TRACKNUMBER" \
-meta:totaltracks="$TRACKTOTAL" \
-meta:disc="$DISCNUMBER" \
-meta:totaldiscs="$DISCTOTAL" \
-meta:genre="$GENRE"

if [ -e "$TMPCVR" ]; then
"$AACTAG" "$fm4a" -add-cover:front:"$TMPCVR"
rm "$TMPCVR"
fi

done
</nowiki></pre>

=== EAC Settings ===

The following settings are examples of what one might use for music and spoken word sources, respectively. Files will be tagged properly using these commands via neroaactag.exe. These commands do run ReplayGain on the files.

This is the syntax for older versions of EAC. For 1.0 beta 2 and above, variables have a changed syntax. See [[http://wiki.hydrogenaudio.org/index.php?title=EAC_and_AAC]]

====320kbps CBR AAC (Forced LC)====
'''Program, including path, used for compression''': C:\Windows\system32\cmd.exe 
'''Additional Command Line Options''': /c ""C:\path\to\neroaacenc.exe" -cbr 320000 -lc -if %s -of %d && "C:\path\to\Neroaactag.exe" %d -meta:artist="%a" -meta:album="%g" -meta:track="%n" -meta:title="%t" -meta:genre="%m" -meta:year="%y""

====Q0.3 VBR AAC (Forced HE)====
'''Program, including path, used for compression''': C:\Windows\system32\cmd.exe 
'''Additional Command Line Options''': /c ""C:\Program Files\Exact Audio Copy\neroaacenc.exe" -q 0.3 -he -if %s -of %d && "C:\Program Files\Exact Audio Copy\Neroaactag.exe" %d -meta:artist="%a" -meta:album="%g" -meta:track="%n" -meta:title="%t" -meta:genre="%m" -meta:year="%y""

=== Foobar ===

The Nero AAC Codec can be used inside Foobar2000's convert function.

== References ==

* [http://www.hydrogenaudio.org/forums/index.php?showtopic=44310 Recommended Settings Sticky]
* [http://www.hydrogenaudio.org/forums/index.php?showtopic=44283 Discussion Thread for the Recommended Setting Sticky]

== External links ==
* [https://en.wikipedia.org/wiki/Nero_AAC_Codec Nero AAC Codec] at Wikipedia

Apple AAC

2023-04-10T22:55:17Z

Logan589XP: Updated version number to 2.79.

Apple AAC

2023-02-27T03:39:08Z

Logan589XP: Updated version number to 2.77.

Foobar2000:Free Encoder Pack

2023-02-25T01:37:40Z

Logan589XP: Updates version numbers of various encoders in addition to the newest Encoder Pack release date.

{{Software Infobox
| name = foobar2000 Free Encoder Pack
| logo =
| screenshot =
| caption =
| developer = Peter Pawlowski
| released = Unknown
| stable_release = 2022-11-30
| stable_release_date = 2022-11-30
| preview_release =
| preview_release_date =
| preview_release_ref =
| operating_system = Windows
| use = Encoder distribution
| license = Various, inc. GPL
}}
The '''Free Encoder Pack''' gives [[foobar2000]] users the ability to utilize additional command-line encoders when converting audio files.

For standard installations of foobar2000, the files are extracted to a new folder in the foobar2000 installation directory called "encoders".

==Encoders==
{| class="wikitable"
|-
! Codec
! Tool
! Version
! Notes
|-
| rowspan="2" | [[AAC]]
| [[fhgaacenc]]
| 20151024
| Requires [[Winamp]] 5.62 or newer
|-
| rowspan="2" | [[QAAC|qaac]]
| rowspan="2" | 2.77
| rowspan="2" | Requires [[iTunes]]
|-
| rowspan="2" | [[Apple Lossless]]
|-
| [[refalac]]
| 1.77
|
|-
| [[FLAC]]
| flac
| 1.4.2
|
|-
| [[MP3]]
| [[LAME|lame]]
| 3.100
|
|-
| [[Musepack]]
| mpcenc
| 1.30.0
|
|-
| [[Ogg Vorbis]]
| [[oggenc|oggenc2]]
| 2.88
| Uses libVorbis 1.3.7
|-
| [[Opus]]
| opusenc
| 0.2.1
| Uses libopus 1.3.1
|-
| [[WavPack]]
| wavpack
| 5.6.0
|
|}
===fhgaacenc===
fhgaacenc allows foobar2000 to use the [[FhG AAC]] codec, which is included with Winamp 5.62–5.666, with the [[foobar2000:Converter|Converter]] component.

{{panel|'''Note:''' The required enc_fhgaac.dll is not included in Winamp 5.8 and later.|color=yellow}}

====Option 1====
* Download and install the relevant version of Winamp.
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

====Option 2====
* Download the relevant version of Winamp, but '''do not''' install it.
* Open the setup exe file as an archive using a tool such as 7-Zip.
* Copy the following files directly to the foobar2000 "encoders" folder:
** {{code|Plugins\enc_fhgaac.dll}}
** {{code|libmp4v2.dll}}
* Restart foobar2000 to use the "AAC (Winamp FhG)" option.

===qaac===
qaac allows foobar2000 to use the [[Apple AAC]] codec from iTunes with the [[foobar2000:Converter|Converter]] component.

====Option 1====
* Download and install iTunes.
* Restart foobar2000 to use the "AAC (Apple)" option.

====Option 2====
* Download iTunes, but '''do not''' install it.
* Download [https://sites.google.com/site/qaacpage/cabinet/makeportable.zip makeportable.zip] and extract "makeportable.cmd" to the same location as the iTunes installer.
* Run "makeportable.cmd" to extract the required files to a folder called "QTfiles" (and "QTfiles64", if the 64-bit version has been downloaded).
* Move the "QTfiles" folder (and "QTfiles64" folder, if it exists) to the foobar2000 "encoders" folder.
* Restart foobar2000 to use the "AAC (Apple)" option.

==External links==
* {{foobar2000|https://www.foobar2000.org/encoderpack|Free Encoder Pack}}
* [https://github.com/tmkk/fhgaacenc/ fhgaacenc] on GitHub
* [https://github.com/nu774/qaac qaac] on GitHub
* [https://github.com/wieslawsoltes/BatchEncoder/wiki/Tutorial-Using-qaac-without-iTunes Using qaac without iTunes] on GitHub

[[Category:foobar2000]]