Hydrogenaudio Knowledgebase - User contributions [en]

Rockbox

2016-11-08T16:42:06Z

178.95.132.4: added References

{{featured}}
[[Image:Rockboxlogo.png|right]]

'''Rockbox''' is a [[GPL]]-compliant [[open source]] operating system for portable digital audio players (DAPs). The Rockbox Project began in 2002 and was first implemented on the [[Archos]] Studio DAP because of owner frustration with severe limitations in the manufacturer-supplied user interface and device operations.

Rockbox can completely replace the host device's operating system firmware and has matured to become an extensible, flexible platform that provides a plug-in architecture for adding PDA functionality, applications, utilities, and games, and has also managed to retrofit video playback functionality onto DAPs first released in mid-2000. Recently, Rockbox now includes a voice-driven user-interface suitable for operation by blind and visually impaired users.

Although Rockbox's official title is "Rockbox: Open Source Jukebox Firmware", in many instances it is not actually installed to (or run from) flash memory. Instead a minimal bootloader is installed in the supported device's flash which is capable of either loading Rockbox from the hard disk or, alternately, the original factory firmware.

== Codecs ==

Rockbox on software decoding platforms (non-Archos) supports playback of eleven [[lossy compression|lossy]] codecs (depending on how one counts), five [[lossless data compression|lossless]], two uncompressed and six miscellaneous formats.<ref>{{cite web|title=Rockbox Supported audio formats|url=http://download.rockbox.org/daily/manual/rockbox-sansaclipplus/rockbox-buildap2.html#x17-335000B.1|work=Rockbox Manual}}</ref> This makes a conservative total of 25 supported audio formats, although a few of them do not operate in realtime on all platforms. Extensive work has gone into optimizing each codec, with FLAC, Ogg, WMA, APE and WMA Pro among the fastest known implementations for those formats.<ref>{{cite web|url=http://www.hydrogenaudio.org/forums/index.php?showtopic=82125&view=findpost&p=716976 |title=Codec performance comparison – Hydrogenaudio Forums |publisher=Hydrogenaudio.org |date= |accessdate=2011-03-12}}</ref>

=== Lossy formats ===

* MPEG audio layers I-III ([[MP3]]/[[MPEG-1 Audio Layer II|MP2]]/[[MPEG-1 Audio Layer I|MP1]])
* [[Vorbis|Ogg Vorbis]]
* [[Advanced Audio Coding|MPEG-4 AAC]](-LC/HE/HEv2 profiles) (in [[MPEG-4 Part 14|MP4]] or [[RealMedia|RM]] containers)
* [[Musepack]]
* [[Dolby Digital|AC3]] (raw or [[RealMedia|RM]] container)
* [[Windows Media Audio|WMA Standard]]
* [[Windows Media Audio|WMA Professional]]
* [[Speex]]
* [[Cook Codec|Cook]]
* [[Adaptive Transform Acoustic Coding#ATRAC3 (LP2 and LP4 Modes)|ATRAC3]]
* The lossy portion of [[WavPack]] hybrid files
* [[Opus]]

=== Lossless formats ===

* [[Free Lossless Audio Codec|FLAC]]
* [[WavPack]]
* [[Shorten]]
* [[Apple Lossless]]
* [[Monkey's Audio]]
* [[TTA (codec)|TTA]]

=== Uncompressed formats ===

* Intel-style [[WAV]]
* Apple [[Audio Interchange File Format|AIFF]]
Together they include over a dozen different [[Pulse-code modulation|PCM]] and [[Adaptive DPCM|ADPCM]] formats.

== Rockbox features ==

Beside the ability of playing and recording audio files, Rockbox offers many playback enhancements that other firmware packages may not have implemented yet. Listed below are a handful of these features.

* [[Gapless playback]]<ref>{{cite web|title=Codec Featureset|url=http://download.rockbox.org/daily/manual/rockbox-sansaclipplus/rockbox-buildap2.html#x17-339000B.1.4|work=Rockbox Manual|accessdate=22 May 2011}}</ref>
* [[crossfader|Crossfading]]<ref>{{cite web|title=Crossfade|url=http://download.rockbox.org/daily/manual/rockbox-sansaclipplus/rockbox-buildch7.html#x10-1220007.7|work=Rockbox Manual|accessdate=22 May 2011}}</ref>
* [[ReplayGain]]<ref name="soft_decode">Software decoding targets only</ref>
* 5 band fully parametric [[equalization (audio)|equalizer]]<ref name="soft_decode" />
* Variable speed decoding with pitch correction<ref>{{cite web|title=Pitch|url=http://download.rockbox.org/daily/manual/rockbox-sansaclipplus/rockbox-buildch4.html#x7-630004.3.3|work=Rockbox Manual|accessdate=22 May 2011}}</ref>
* [[Crossfeed]]<ref name="soft_decode" />
* OTF ("on the fly") playlists
* True random shuffle (fresh randomly shuffled list every time)
* Custom [[Theme (computing)|UI themes]]
* Dynamic Playlists (queue files to play next, or in other parts of a dynamic playlist)
* Stereo recording to WAV/AIFF/WavPack (lossless) and MP3<ref>MP3, WavPack and AIFF are available on non-Archos devices. Multiple sample rates and bitrates available (hardware-dependent).</ref><ref>{{cite web|title=Recording|url=http://download.rockbox.org/daily/manual/rockbox-sansaclipplus/rockbox-buildch10.html#x13-14900010|work=Rockbox Manual|accessdate=22 May 2011}}</ref> (supporting devices)
* [[FM broadcasting|FM radio]], including FM recording (supporting devices)
* Remote control (supporting devices)
* Digital [[S/PDIF]] input/output (supporting devices)
* [[Last.fm]] support (even on players lacking [[Real-time clock|RTC]])
* [[cue sheet (computing)|Cue sheet]] support
* Changeable selector bar
* Album art<ref>{{cite web|url=http://www.rockbox.org/twiki/bin/view/Main/AlbumArt |title=Some limitations. Details at Rockbox Wiki |publisher=Rockbox.org |date= |accessdate=2011-03-12}}</ref>
* Sleep timer

== External links ==
* [http://www.rockbox.org/ The Rockbox Project]

== References ==

<references/>

<span style="color:green;">''~ Text taken from [http://en.wikipedia.org/wiki/Rockbox Wikipedia entry for Rockbox]''</span>

[[Category:Firmware]]

MPC Encoder Functions

2016-10-19T08:26:37Z

178.95.132.4: /* Quality oriented encoder functions */ fixing formatting again...

==Quality oriented encoder functions==

; --ms x ''(x can be: 0, 1, 2)''
: Sets [[Mid-side stereo]] mode (channel coupling): 0 (off), 1 (on) or 2 (enhanced): 0 means that there's no channel coupling, 2 means channel coupling is more cautious. 1 means, channel coupling is not so cautious which may result joint stereo artifacts.

: M/S-coding calculates a "mid"-channel by addition of left and right channel (l+r)/2 and a "side"-channel (l-r)/2. With more mono-like signals one can use less [[bitrate]] to encode the side-channel, so that the overall bitrate will be less than encoding the left and right channel. If the psychoacoustics work well, there is no audible difference between m/s- coded or l/r-coded files. Mid/Side coding in MPC is [[subband]] selective, broadband 0-22khz is divided into 32 subbands. [[Psychoacoustic|Psychoacoustics]] calculates for each subband if mid/side coding should be used or not. This is different than in [[MP3]] encoding, where the full frame will be either m/s coded or l/r (true stereo) coded, so MP3 mid/side coding is more likely to cause audible artifacts, unless tweaked to be very cautious (like nssafejoint in lame encoder).

; --cvd x ''(x can be: 0, 1)''
: Sets clearvoicedetection either off (0) or on (1 default).
: CVD is able to detect voice-like signals to give a higher quality with voices or sounds with harmonic spectra. It uses special analysis to detect [[harmonics]] with varying base [[frequency]] - the "normal" psychoacoustics are not able to detect such signals and will add audible noise to these signals.

; --bw x ''(x can be: 0 to 22500)''
: Defines the max frequency [[bandwidth]] which can be encoded (actual frequency response depends also on [[LTQ|ATM]]/[[ATH]]). Basically acts like a [[lowpass|lowpass filter]].

; --ltq x ''(x can be: iso, ank, fil)''
: Level threshold in quiet (also called as ATH) is a threshold or hearing curve. This is the sound pressure level (spl in db) below which the human hearing of most people is unable to perceive a sine-tone.

; --ltq_max x ''(x can be: -99 to 99; recommended: 60 to 99)''
: Maximum level for ltq, in [[dB]]. default is 83.

; --ltq_gain x ''(x can be: -99 to 99; recommended: -12 to 5)''
: Adds offset of x db to chosen ltq. If you use negative number, you can make the hearing curve more sensitive (for more sensitive hearing), but it increases [[bitrate]]. If you use positive number, less bits will be needed.

; --ltq_var x ''(x can be: 0, 1)''
: Adaptive threshold in quiet. default is 1.

; --minSMR x ''(x can be: 0 to 3; recommended: 0)''
: Sets the minimum smr (signal to mask ratio) over full BandWidth. The higher the smr the higher the quality and bitrate. Setting -minSMR over 0 db will result in full BandWidth encoding, like in insane profile.

; --tmpmask x ''(x can be: 0, 1)''
: Sets post [[masking]] on or off. Temporal postmasking saves a few kbit/s because the human hearing has to "relax" after a sound event, so that the encoder can put a bit more distortion to the signal during this time (saves bits).

; --nmt x ''(x can be: 0 to 99; recommended: 6 to 16)''
: Sets minimum smr (signal to mask ratio) for pure noisy sound. MPC encoder calculates a masking threshold. Noisy sound has high masking ratio. [[Subband]] coder like MPC works by adding noise ([[quantization]] error) to each subband. You can increase the quantization resolution (less quantization noise, higher [[bitrate]]) by raising smr.

: This noise should be of course below the masking threshold, so that it would be inaudible. Sometimes quantization noise however is not inaudible, because [[tonality]] estimation (which calculates the tonality and "noisiness" of sound) may conclude that a noisy sound is more noisier than it really is. This will mean that the masking threshold will be higher than it should be. Encoder concludes that more noise can be masked than really can, and this will result audible noise (distortion). This happens because quantization resolution (bitrate) is lower than it should be in order for the noise to be inaudible. But, you can compensate this by raising smr for pure noisy sound (nmt). It will increase the quantization resolution for noisy sound.

; --tmn x ''(x can be: 0 to 99; recommended: 22 to 32)''
: Sets minimum smr for pure sinusoidal sound. Sinusoidal sound is very tonal (not noisy). This means that it does not have much masking capability. Quantization resolution (bitrate) must be high enough so that tonal sound is encoded without audible noise. Of course in normal music there is both noisy and tonal sound, so the masking threshold will be calculated accordingly. Also different resolutions of quantization can be assigned to the different frequency regions.

; --ans x ''(x can be: 0 to 5; recommended: 5)''
: Adaptive noise shaping order. 0 means off, 1 to 5 means on. Default is 5.

;--shortthr x ''(x can be for example: 5)''
: Short fft threshold. default is 5.

; --transdet x ''(x can be for example: 100)''
: Slewrate for [[transient]] detection. default is 100.

; --minval x ''(x can be: 0, 1, 2)''
: Method for calculating minval. 0: old method, 1: Buschmann method, 2: Klemm method.

; --xlevel
: Use alternative [[filterbank]] [[clipping]] solving strategy. Not thoroughly tested.

; --scale x ''(x can be: 0.00 to 1.00; recommended 0.850 to 1.00)''
: Method to overcome [[clipping]] of the encoded file. In the event that clipping occurs, the encoder (mppenc) will display a warning message and recommend a numerical value to process the audio so that clipping may cease. The numerical value is a percentage to the extent that the audio will be processed (1.0 = 100%, 0.85 = 85%). "--scale 1.00" means no additional clipping processing will be done - this is the desired situation.

; --fadeshape x ''(x can be: 1 to 99; recommended: 1 to 10)''
: Sets the fading scheme used. "Small values are first fast fading then slow fading. Large values are the opposite." - FrankKlemm?

; --fadein x / --fadeout x ''(x can be: 0 to 99; recommended: 0 to 5)''
: The encoder (mppenc) offers simple audio processing via fading. "X" indicates the number of seconds to which the encoder will fade the music. Useful for encoding live recordings when the whole recording will not be used (e.g. tracklisting, playlists, seamless listening, shuffled music, etc.).

; --start x / --skip x ''(x can be: 0 to 99; recommended: 0 to 5)''
: Sets the number of seconds that the encoder will NOT PROCESS. (i.e. "--start 4" will skip the first 4 seconds of the source file and the encoder will begin encoding on the fourth second). "Start" and "skip" do the same thing.

MPC Encoder Functions

2016-10-19T08:16:55Z

178.95.132.4: /* Quality oriented encoder functions */ better formating

==Quality oriented encoder functions==

; --ms x ''(x can be: 0, 1, 2)''
: Sets [[Mid-side stereo]] mode (channel coupling): 0 (off), 1 (on) or 2 (enhanced): 0 means that there's no channel coupling, 2 means channel coupling is more cautious. 1 means, channel coupling is not so cautious which may result joint stereo artifacts.

: M/S-coding calculates a "mid"-channel by addition of left and right channel (l+r)/2 and a "side"-channel (l-r)/2. With more mono-like signals one can use less [[bitrate]] to encode the side-channel, so that the overall bitrate will be less than encoding the left and right channel. If the psychoacoustics work well, there is no audible difference between m/s- coded or l/r-coded files. Mid/Side coding in MPC is [[subband]] selective, broadband 0-22khz is divided into 32 subbands. [[Psychoacoustic|Psychoacoustics]] calculates for each subband if mid/side coding should be used or not. This is different than in [[MP3]] encoding, where the full frame will be either m/s coded or l/r (true stereo) coded, so MP3 mid/side coding is more likely to cause audible artifacts, unless tweaked to be very cautious (like nssafejoint in lame encoder).

; --cvd x ''(x can be: 0, 1)''
: Sets clearvoicedetection either off (0) or on (1 default).
: CVD is able to detect voice-like signals to give a higher quality with voices or sounds with harmonic spectra. It uses special analysis to detect [[harmonics]] with varying base [[frequency]] - the "normal" psychoacoustics are not able to detect such signals and will add audible noise to these signals.

; --bw x ''(x can be: 0 to 22500)''
: Defines the max frequency [[bandwidth]] which can be encoded (actual frequency response depends also on [[LTQ|ATM]]/[[ATH]]). Basically acts like a [[lowpass|lowpass filter]].

; --ltq x ''(x can be: iso, ank, fil)''
: Level threshold in quiet (also called as ATH) is a threshold or hearing curve. This is the sound pressure level (spl in db) below which the human hearing of most people is unable to perceive a sine-tone.

; --ltq_max x ''(x can be: -99 to 99; recommended: 60 to 99)''
: Maximum level for ltq, in [[dB]]. default is 83. - --ltq_gain x (x can be: -99 to 99; recommended: -12 to 5)
: Adds offset of x db to chosen ltq. If you use negative number, you can make the hearing curve more sensitive (for more sensitive hearing), but it increases [[bitrate]]. If you use positive number, less bits will be needed.

; --ltq_var x ''(x can be: 0, 1)''
: Adaptive threshold in quiet. default is 1.

; --minSMR x ''(x can be: 0 to 3; recommended: 0)''
: Sets the minimum smr (signal to mask ratio) over full BandWidth. The higher the smr the higher the quality and bitrate. Setting -minSMR over 0 db will result in full BandWidth encoding, like in insane profile.

; --tmpmask x ''(x can be: 0, 1)''
: Sets post [[masking]] on or off. Temporal postmasking saves a few kbit/s because the human hearing has to "relax" after a sound event, so that the encoder can put a bit more distortion to the signal during this time (saves bits).

; --nmt x ''(x can be: 0 to 99; recommended: 6 to 16)''
: Sets minimum smr (signal to mask ratio) for pure noisy sound. MPC encoder calculates a masking threshold. Noisy sound has high masking ratio. [[Subband]] coder like MPC works by adding noise ([[quantization]] error) to each subband. You can increase the quantization resolution (less quantization noise, higher [[bitrate]]) by raising smr.

: This noise should be of course below the masking threshold, so that it would be inaudible. Sometimes quantization noise however is not inaudible, because [[tonality]] estimation (which calculates the tonality and "noisiness" of sound) may conclude that a noisy sound is more noisier than it really is. This will mean that the masking threshold will be higher than it should be. Encoder concludes that more noise can be masked than really can, and this will result audible noise (distortion). This happens because quantization resolution (bitrate) is lower than it should be in order for the noise to be inaudible. But, you can compensate this by raising smr for pure noisy sound (nmt). It will increase the quantization resolution for noisy sound.

; --tmn x ''(x can be: 0 to 99; recommended: 22 to 32)''
: Sets minimum smr for pure sinusoidal sound. Sinusoidal sound is very tonal (not noisy). This means that it does not have much masking capability. Quantization resolution (bitrate) must be high enough so that tonal sound is encoded without audible noise. Of course in normal music there is both noisy and tonal sound, so the masking threshold will be calculated accordingly. Also different resolutions of quantization can be assigned to the different frequency regions.

; --ans x ''(x can be: 0 to 5; recommended: 5)''
: Adaptive noise shaping order. 0 means off, 1 to 5 means on. Default is 5.

;--shortthr x ''(x can be for example: 5)''
: Short fft threshold. default is 5.

; --transdet x ''(x can be for example: 100)''
: Slewrate for [[transient]] detection. default is 100.

; --minval x ''(x can be: 0, 1, 2)''
: Method for calculating minval. 0: old method, 1: Buschmann method, 2: Klemm method.

; --xlevel
: Use alternative [[filterbank]] [[clipping]] solving strategy. Not thoroughly tested.

; --scale x ''(x can be: 0.00 to 1.00; recommended 0.850 to 1.00)''
: Method to overcome [[clipping]] of the encoded file. In the event that clipping occurs, the encoder (mppenc) will display a warning message and recommend a numerical value to process the audio so that clipping may cease. The numerical value is a percentage to the extent that the audio will be processed (1.0 = 100%, 0.85 = 85%). "--scale 1.00" means no additional clipping processing will be done - this is the desired situation.

; --fadeshape x ''(x can be: 1 to 99; recommended: 1 to 10)''
: Sets the fading scheme used. "Small values are first fast fading then slow fading. Large values are the opposite." - FrankKlemm?

; --fadein x / --fadeout x ''(x can be: 0 to 99; recommended: 0 to 5)''
: The encoder (mppenc) offers simple audio processing via fading. "X" indicates the number of seconds to which the encoder will fade the music. Useful for encoding live recordings when the whole recording will not be used (e.g. tracklisting, playlists, seamless listening, shuffled music, etc.).

; --start x / --skip x ''(x can be: 0 to 99; recommended: 0 to 5)''
: Sets the number of seconds that the encoder will NOT PROCESS. (i.e. "--start 4" will skip the first 4 seconds of the source file and the encoder will begin encoding on the fourth second). "Start" and "skip" do the same thing.

Lossless comparison

2016-10-16T07:29:01Z

178.95.132.4: /* MLP/Dolby TrueHD */ it -> is

The '''lossless comparison page''' aims to gather information about lossless codecs available so users can make an informed decision as to what lossless codec to choose for their needs.

== Introduction ==
Given the enormous amount of [[lossless]] audio compressor choices available, it is a very difficult task to choose the one most suited for each person's needs. Some people only take into consideration compression performance when choosing a codec, but as the following table and article shows, there are several other features worth taking into consideration when making a choice.

For example, users wanting good multiplatform compatibility and robustness (e.g., people sharing live recordings) would favour [[WavPack]] or [[FLAC]]. Another user, looking for the very highest compression available, would go with [[OptimFROG]]. Someone wanting portable support would use [[FLAC]] or [[ALAC]], and so on. En fin, this is not a matter worth getting too worked up about. If you later find out the codec you chose isn't the best for your needs, you can just transcompress to another format, without risk of losing quality.

'''Note:''' for latest comparison of lossless compression, scroll down to the [[Lossless comparison#External links|Links section of this page]].

== Comparison Table ==


{| class="wikitable" cellspacing="2"
!width="120px"|'''Features'''
! width="90px" | [[#Free Lossless Audio Codec (FLAC)|FLAC]]
! width="90px" | [[#Apple Lossless Audio Codec (ALAC)|ALAC]]
! width="90px" | [[#WavPack (WV)|WavPack]]
! width="90px" | [[#Tom's_verlustfreier_Audiokompressor (TAK)|TAK]]
! width="90px" | [[#Monkey's_Audio (APE)|Monkey's]]
! width="90px" | [[#Windows Media Audio Lossless (WMAL)|WMAL]]
! width="90px" | [[#OptimFROG (OFR)|OptimFROG]]
! width="90px" | [[#True Audio (TTA)|TTA]]
|- 
| Encoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #FFCC66" | slow
| style="background: #00FF00" | very fast
|- 
| Decoding speed{{ref label|speed|A|A}}
| style="background: #00FF00" | very fast
| style="background: #CCFFCC" | fast
| style="background: #CCFFCC" | fast
| style="background: #00FF00" | very fast
| style="background: #FFCC66" | slow
| style="background: #FFCC66" | average
| style="background: #FF9900" | very slow
| style="background: #CCFFCC" | fast
|- 
| Compression{{ref label|speed|A|A}}{{ref label|comp|B|B}}
| style="background: #CCFFCC" | 57.0%
| style="background: #CCFFCC" | 57.8%
| style="background: #CCFFCC" | 57.1%
| style="background: #00FF00" | 56.0%
| style="background: #00FF00" | 55.1%
| style="background: #FFCC66" | 58.4%
| style="background: #00FF00" | 54.6%
| style="background: #CCFFCC" | 56.6%
|-
| # presets
| style="background: #CCFFCC" | 9
| style="background: #CCFFCC" | 2
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 5
| style="background: #CCFFCC" | 1
| style="background: #CCFFCC" | > 10
| style="background: #CCFFCC" | 1
|-
| Error handling{{ref label|error|C|C}}
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | yes{{ref label|error_ape|D|D}}
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Tagging
| style="background: #00FF00" | Vorbis tags
| style="background: #00FF00" | iTunes
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | APEv2
| style="background: #00FF00" | ASF
| style="background: #CCFFCC" | ID3/APEv2
| style="background: #00FF00" | ID3v1/2 or APEv2
|-
| Hardware support
| style="background: #00FF00" | very good
| style="background: #00FF00" | very good
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
| style="background: #FFCC66" | limited
| style="background: #FF9900" | no
| style="background: #FFCC66" | limited
|-
| Software support
| style="background: #00FF00" | very good
| style="background: #00FF00" | very good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
| style="background: #CCFFCC" | good
| style="background: #FFCC66" | average
| style="background: #CCFFCC" | good
|-
| Hybrid/lossy
| style="background: #CCFFCC" | LossyWAV
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #CCFFCC" | LossyWAV
| style="background: #FF9900" | no
| style="background: #CCFFCC" | LossyWAV
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| RIFF chunks
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
|-
| Streaming
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
|-
| Open source
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| Multichannel
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
| style="background: #FF9900" | no
| style="background: #00FF00" | yes
|-
| OS support
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Wine
| style="background: #00FF00" | All
| style="background: #CCFFCC" | Win/Mac
| style="background: #00FF00" | Win/Mac/Linux
| style="background: #00FF00" | All
|}

{|
|-
|{{note label|speed|A|A}} Speed and Compression are based on '''each encoder's default settings''' and taken from [http://www.audiograaf.nl/downloads.html this comparison].
|-
|{{note label|comp|B|B}} The Compression ratio is compressed size/uncompressed size * 100. So, lower is better.
|-
|{{note label|error|C|C}} Error handling means that a codec can detect a corruption (flipped bit) in a file and warn the user about it, but it will still decode the rest of the file.
|-
|{{note label|error_ape|D|D}} The official Monkey's Audio decoder does not support decoding through errors, but this may be achieved with FFmpeg or Winamp, though likely not, when the "Insane" preset is used.

== Codecs ==

These are the most popular lossless codecs, in alphabetical order:

=== Apple Lossless Audio Codec (ALAC) ===
https://alac.macosforge.org/trac

[[ALAC]] is a codec developed by Apple for usage in [[Apple iPod|iPod]] and AirPort Express.

'''ALAC pros'''
* [[Open source]] (encoding and decoding via FFmpeg and [[CueTools|CUETools]], decoding only via [http://craz.net/programs/itunes/alac.html a standalone decoder])
* Fast encoding
* Fast decoding
* Hardware support ([[Apple iPod|iPod]], AirPort Express)
* Software support (iTunes, Quicktime)
* Independent encoder implementation available: ffmpeg
* Streaming support
* Tagging support (QT tags)
* Supports [[multichannel]] audio and [[high resolution]]s
* Used by a few online stores

''' ALAC cons '''
* Limited software support
* No error detection/robustness<ref>[http://www.hydrogenaud.io/forums/index.php?s=&showtopic=33226&view=findpost&p=862031 HA forum post discussing ALAC robustness]</ref>
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Not very efficient

''' ALAC Other features '''
* Fits in the [[MP4]] container

=== Free Lossless Audio Codec (FLAC) ===
https://xiph.org/flac/

[[FLAC]] is a lossless codec developed by Josh Coalson. It's part of the Xiph multimedia portfolio, along with [[Opus]], [[Ogg]], [[Vorbis]], [[Speex]] and [[Theora]].

''' FLAC pros '''
* [[Open source]]
* Very fast decoding
* Very fast encoding
* Very good hardware support (Android, Marantz, Sonos, [http://xiph.org/flac/links.html many others])
* Very good software support
* Independent encoder implementations available: flake/ffmpeg, FLACCL
* Error robustness
* Streaming support
* Supports [[multichannel]] audio and [[high resolution]]s
* Tagging support (FLAC tags)
* Supports [[RIFF]] chunks
* Pipe support
* Used by a few [http://xiph.org/flac/links.html#music online stores]

''' FLAC cons '''
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Does not handle 32-bit float and there is no encoder that can render to 32-bit integer

''' FLAC Other features '''
* Supports embedded CUE sheets (with [http://flac.sourceforge.net/faq.html#general__no_cuesheet_tags limitations])
* Includes MD5 hashes for quick integrity checking as standard
* Fits the [[Ogg]] and [[Matroska]] containers

=== Monkey's Audio (APE) ===
http://www.monkeysaudio.com/

[[Monkey's Audio]] is a very efficient lossless compressor developed by Matt Ashland.

''' APE pros '''
* High compression
* Fast encoding
* Good software support
* Simple and user friendly. Official GUI provided.
* Java version (multiplatform)
* Error robustness/decoding up to -c3000 (High compression)<ref>http://www.hydrogenaud.io/forums/index.php?showtopic=98984&st=0&p=821420&#entry821420</ref>
* Tagging support ([[ID3v1]], [[APE tags]])
* [[High resolution]] audio support
* Supports [[RIFF]] chunks (only in the GUI encoder)
* Pipe support (only in a [http://www.etree.org/shnutils/shntool/ special] version)

''' APE cons '''
* Problematic license (source provided, no modification or redistribution rights)
* Slow decoding
* No [[multichannel]] support
* No hybrid/lossy mode (and not [[LossyWAV]] compatible)
* Limited hardware support (Rockbox, some Cowon players); poor battery life due to complicated decoding (see [http://www.rockbox.org/wiki/SoundCodecMonkeysAudio MP3 player benchmarks])
* Higher compression levels are extremely CPU intensive

''' APE Other features '''
* Includes MD5 hashes for quick integrity checking
* Supports APL image link files (similar to CUE sheets)

=== OptimFROG (OFR) ===
http://www.losslessaudio.org/

[[OptimFROG]] is a lossless format developed by Florin Ghido to become the champion in audio compression.

''' OFR pros '''
* Very high compression
* Good software support
* Error robustness
* Streaming support
* Supports [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3]], [[APE tags]])
* Supports [[RIFF]] chunks

''' OFR cons '''
* Closed source
* No [[multichannel]] audio support
* No hardware support
* Very slow decoding
* Slow encoding
* More than one tagging method allowed (ambiguity possible)

''' OFR Other features '''
* Supports 32bit float streams
* Includes MD5 hashes for quick integrity checking

=== Tom's verlustfreier Audiokompressor (TAK) ===
http://www.thbeck.de/Tak/Tak.html

[[TAK]] is a lossless codec developed by Thomas Becker.

''' TAK pros '''
* Very fast decoding
* Very fast encoding
* Very high efficiency
* Error robust
* Supports multichannel audio and high resolutions
* Tagging support
* Supports RIFF chunks
* Pipe support
* Streamable

''' TAK cons '''
* Closed source
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* No hardware support
* Average software support
* Doesn't support Unicode (yet)

''' TAK Other features '''
* Optional MD5 checksum

=== True Audio (TTA) ===
http://tta.tausoft.org/

[[TTA]] is a lossless codec developed by a international team of programmers.

''' TTA pros '''
* [[Open source]]
* Supports [[multichannel]] audio and [[high resolution]]s
* Tagging support ([[ID3]]v1, ID3v2 or [[APEv2]])
* Embedded CUE sheets support
* Error robustness
* Pipe support
* Average compression
* Fast encoding/decoding
* Symmetric algorithm
* Password protection
* Ultra low latency

''' TTA cons '''
* No hybrid/lossy mode
* Doesn't support [[RIFF]] chunks
* Limited hardware support

''' TTA Other features '''
* Fits the [[Matroska]] container
* Password protection

=== WavPack (WV) ===
http://www.wavpack.com/

[[WavPack]] is a fast and featureful lossless codec developed by David Bryant.

''' WV pros '''
* [[Open source]]
* Fast decoding
* Very fast encoding
* Good efficiency
* Error robustness
* Streaming support
* Supports [[multichannel]] audio and [[high resolution]]s
* Hybrid/lossy mode
* Tagging support ([[ID3v1]], [[APE tags]])
* Supports [[RIFF]] chunks
* Ability to create self extracting files for Win32 platform
* Pipe support
* Good software support

''' WV cons '''
* Limited hardware player support ([http://www.rockbox.org/ RockBox])
* More than one tagging method allowed (ambiguity possible)

''' WV Other features '''
* Supports 32bit float streams
* Supports embedded CUE sheets
* Includes MD5 hashes for quick integrity checking
* Can encode in both symmetrical and asymmetrical modes.
* Fits the [[Matroska]] container

=== Windows Media Audio Lossless (WMAL) ===
https://msdn.microsoft.com/en-us/library/ff819508(v=vs.85).aspx

WMA Lossless is the lossless codec developed by Microsoft to be featured in their Windows Media codec portfolio.

''' WMAL pros '''
* Streaming support
* Very good software support
* Hardware support (Microsoft Zune, [http://en.wikipedia.org/wiki/Gigabeat Gigabeat V and S line from Toshiba])
* Supports [[multichannel]] audio and [[high resolution]]s.
* Tagging support (proprietary)
* Pipe support

''' WMAL cons '''
* Rather low efficiency
* Closed source
* No hybrid/lossy mode (but is [[LossyWAV]] compatible)
* Doesn't support [[RIFF]] chunks

''' WMAL Other features '''
* Fits the [[ASF]] container

=== Other Formats ===
Aside from the formats mentioned above, there are in fact quite a lot of other lossless formats. To keep the table and list brief and readable, a few formats have not been mentioned.

====DTS-HD Master Audio====
Similar to the MPEG-4 SLS format, this format has a core track in an older, more widely supported format, DTS. This core lossy track is made lossless by a secondary track with correction data. It is an optional codec in Blu-ray implementations. Its main use is surround sound encoding, and as is the case with MLP, the price of the encoder ensures it is only used in mastering of Blu-ray discs.

====LA====
http://www.lossless-audio.com/

LA features an extremely high compression (on par with OptimFrog highest modes, but a bit faster), but it hasn't been updated for more than 10 years. Furthermore, backward compatibility is not guaranteed, so using it for archiving might pose a few problems. It isn't able to cope with file corruption either, software support is very limited and isn't open source.

====MLP/Dolby TrueHD====
The [[MLP|MLP codec]] (of which the mathematical basis was used in Dolby TrueHD) is the codec used for DVD-Audio. It was mandatory in any HD-DVD implementation and optional for Blu-Ray in it's Dolby TrueHD form. It is known to support the 'wasted bits' scheme used in LossyWAV. As encoders are very expensive, its use outside DVD/Blu-ray mastering environments is non-existent. Its main use is encoding surround sound data.

====MPEG-4 ALS====
MPEG-4 ALS is the successor to LPAC, which it was based on. It has been as a ISO standard and there is a reference encoder/decoder, but like TTA, it does not have features that make it stand out from other codecs, nor backing by a large organisation, so it hasn't much software and no hardware support.

====MPEG-4 SLS====
MPEG-4 SLS is a special codec, having a AAC core track and a 'correction track'. Also known as HD-AAC, SLS stands for Scalable to Lossless. However, there is to date still no affordable software to play, encode or decode (the lossless part of) SLS files.

====Shorten====
http://www.etree.org/shncom.html

Shorten was one of the first widely-used lossless formats, and it still occasionally found on the internet, especially in archives, for example etree.org. It is quite fast in both encoding and decoding, but doesn't compress very much. Furthermore, seeking has a troubled past as well as tagging. It is considered obsolete.

====Real Lossless====
Part of the Real codec suite, Real Lossless too hasn't any very special features that make it stand out. Just like WMA Lossless and Apple Lossless, it was created to fit in a codec suite, but unlike WMA Lossless and Apple Lossless, there is no hardware support and software support is limited. Compression is on par with most other codecs, but it is rather slow to encode.

====Oddball formats====
There are a few archaic formats of which encoders and decoders are hard to get by. Most of those would have disappeared by now, but some of them are being preserved for posterity at [[User:Rjamorim|rjamorim]]'s

* Advanced Digital Audio (ADA)
* [http://www.logarithmic.net/pfh/bonk Bonk]
* Marian's a-Pac
* AudioZip
* Dakx WAV
* Entis Lab MIO
* LiteWave
* LPAC
* Pegasus SPS
* [http://www.free-codecs.com/download/rk_audio_compressor.htm RK Audio (RKAU)]
* Ogg Squish
* Sonarc
* VocPack
* [http://www.firstpr.com.au/audiocomp/lossless/wavarc/ WavArc]
* [http://www.firstpr.com.au/audiocomp/lossless/WaveZip/ WaveZip]/MUSICompress

== See also ==
* [[Lossless]]

== External links ==
''' Other lossless compressions comparisons '''
''Sorted based on last '''update''' date.''

* [http://www.audiograaf.nl/downloads.html Martijn van Beurden's comparison] - tries to compare all codecs and settings with a balanced pool of music (last updated 2015-01-05)
* [http://www.squeezechart.com/audio.html Squeezechart audio] - tests as much codecs as possible, but not all their settings and with a limited test corpus (last updated 2013-10-31)
* [http://synthetic-soul.co.uk/comparison/lossless/index.asp Synthetic Soul's comparison] (last update 2007-07-28)
* <s>Johan De Bock's speed oriented comparison</s> - best choices speedwise are indicated in green, mostly electronic music (last updated 2006-07-22)
* <s>Hans Heijden's</s> -- used as reference to build the table (last updated 2006-07-07)
* <s>Josef Pohm's comparison, hosted by Synthetic Soul</s> (last update 2006-05-29)
* [http://www.bobulous.org.uk/misc/lossless_audio_2006.html Bobulous' lossless audio comparison] — a look at six lossless formats in terms of speed and file size (last updated 2006-05-22)
* <s>Jhan De Bock's size oriented comparison</s> - aimed only at the maximum compression setting for each codec (based on a somewhat limited set of samples, however) (last updated 2006-05-19)
* <s>Gruboolez'</s> -- comparing only classical music (last updated 2005-02-27)
* <s>Speek's</s> (last updated 2005-02-07)
*[http://www.firstpr.com.au/audiocomp/lossless/ Lossless Compression of Audio] Much information about oddball formats including comparison of them. (last updated 2005-10-21)

''' More on lossless compressions '''
* [http://web.archive.org/web/20080731103800/http://www.losslessaudioblog.com/ The Lossless Audio Blog], retrieved from archive.org - by windmiller, is a reliable and complete source of news about lossless compression.
* Go to the [http://www.hydrogenaudio.org/forums/index.php?showtopic=33226 Hydrogenaudio thread] to discuss this article.

== References ==

<references/>
[[Category:Guides]]