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format
This is a detailed description of the FLAC format. There is also a
companion document that describes FLAC-to-Ogg mapping .
For a user-oriented overview, see About the FLAC Format
.
*_Table of Contents_*
* Acknowledgments <#acknowledgments>
* Scope <#scope>
* Architecture <#architecture>
* Definitions <#definitions>
* Blocking <#blocking>
* Interchannel Decorrelation <#interchannel>
* Prediction <#prediction>
* Residual Coding <#residualcoding>
* Format <#format_overview>
* FLAC Subset <#subset>
* Specification
o STREAM <#stream>
+ METADATA_BLOCK <#metadata_block>
# METADATA_BLOCK_HEADER <#metadata_block_header>
# METADATA_BLOCK_DATA <#metadata_block_data>
* METADATA_BLOCK_STREAMINFO
<#metadata_block_streaminfo>
* METADATA_BLOCK_PADDING
<#metadata_block_padding>
* METADATA_BLOCK_APPLICATION
<#metadata_block_application>
* METADATA_BLOCK_SEEKTABLE
<#metadata_block_seektable>
o SEEKPOINT <#seekpoint>
* METADATA_BLOCK_VORBIS_COMMENT
<#metadata_block_vorbis_comment>
* METADATA_BLOCK_CUESHEET
<#metadata_block_cuesheet>
o CUESHEET_TRACK <#cuesheet_track>
+ CUESHEET_TRACK_INDEX
<#cuesheet_track_index>
* METADATA_BLOCK_PICTURE
<#metadata_block_picture>
o
+ FRAME <#frame>
# FRAME_HEADER <#frame_header>
# FRAME_FOOTER <#frame_footer>
# SUBFRAME <#subframe>
* SUBFRAME_HEADER <#subframe_header>
* SUBFRAME_CONSTANT <#subframe_constant>
* SUBFRAME_FIXED <#subframe_fixed>
* SUBFRAME_LPC <#subframe_lpc>
* SUBFRAME_VERBATIM <#subframe_verbatim>
o RESIDUAL <#residual>
+ RESIDUAL_CODING_METHOD_PARTITIONED_RICE
<#partitioned_rice>
# RICE_PARTITION
<#rice_partition>
+ RESIDUAL_CODING_METHOD_PARTITIONED_RICE2
<#partitioned_rice2>
# RICE2_PARTITION
<#rice2_partition>
*_Acknowledgments_*
FLAC owes much to the many people who have advanced the audio
compression field so freely. For instance:
* A. J. Robinson for his work
on Shorten
;
his paper is a good starting point on some of the basic methods
used by FLAC. FLAC trivially extends and improves the fixed
predictors, LPC coefficient quantization, and Rice coding used in
Shorten.
* S. W. Golomb and
Robert F. Rice; their universal codes are used by FLAC's entropy
coder.
* N. Levinson and J. Durbin; the reference encoder uses an algorithm
developed and refined by them for determining the LPC coefficients
from the autocorrelation coefficients.
* And of course, Claude Shannon
*_Scope_*
It is a known fact that no algorithm can losslessly compress all
possible input, so most compressors restrict themselves to a useful
domain and try to work as well as possible within that domain. FLAC's
domain is audio data. Though it can losslessly *code* any input, only
certain kinds of input will get smaller. FLAC exploits the fact that
audio data typically has a high degree of sample-to-sample correlation.
Within the audio domain, there are many possible subdomains. For
example: low bitrate speech, high-bitrate multi-channel music, etc. FLAC
itself does not target a specific subdomain but many of the default
parameters of the reference encoder are tuned to CD-quality music data
(i.e. 44.1kHz, 2 channel, 16 bits per sample). The effect of the
encoding parameters on different kinds of audio data will be examined later.
*_Architecture_*
Similar to many audio coders, a FLAC encoder has the following stages:
* Blocking <#blocking>. The input is broken up into many contiguous
blocks. With FLAC, the blocks may vary in size. The optimal size
of the block is usually affected by many factors, including the
sample rate, spectral characteristics over time, etc. Though FLAC
allows the block size to vary within a stream, the reference
encoder uses a fixed block size.
* Interchannel Decorrelation <#interchannel>. In the case of stereo
streams, the encoder will create mid and side signals based on the
average and difference (respectively) of the left and right
channels. The encoder will then pass the best form of the signal
to the next stage.
* Prediction <#prediction>. The block is passed through a prediction
stage where the encoder tries to find a mathematical description
(usually an approximate one) of the signal. This description is
typically much smaller than the raw signal itself. Since the
methods of prediction are known to both the encoder and decoder,
only the parameters of the predictor need be included in the
compressed stream. FLAC currently uses four different classes of
predictors (described in the prediction <#prediction> section),
but the format has reserved space for additional methods. FLAC
allows the class of predictor to change from block to block, or
even within the channels of a block.
* Residual coding <#residualcoding>. If the predictor does not
describe the signal exactly, the difference between the original
signal and the predicted signal (called the error or residual
signal) must be coded losslessy. If the predictor is effective,
the residual signal will require fewer bits per sample than the
original signal. FLAC currently uses only one method for encoding
the residual (see the Residual coding <#residualcoding> section),
but the format has reserved space for additional methods. FLAC
allows the residual coding method to change from block to block,
or even within the channels of a block.
In addition, FLAC specifies a metadata system, which allows arbitrary
information about the stream to be included at the beginning of the stream.
*_Definitions_*
Many terms like "block" and "frame" are used to mean different things in
differenct encoding schemes. For example, a frame in MP3 corresponds to
many samples across several channels, whereas an S/PDIF frame represents
just one sample for each channel. The definitions we use for FLAC
follow. Note that when we talk about blocks and subblocks we are
referring to the raw unencoded audio data that is the input to the
encoder, and when we talk about frames and subframes, we are referring
to the FLAC-encoded data.
* *Block*: One or more audio samples that span several channels.
* *Subblock*: One or more audio samples within a channel. So a block
contains one subblock for each channel, and all subblocks contain
the same number of samples.
* *Blocksize*: The number of samples in any of a block's subblocks.
For example, a one second block sampled at 44.1KHz has a blocksize
of 44100, regardless of the number of channels.
* *Frame*: A frame header plus one or more subframes.
* *Subframe*: A subframe header plus one or more encoded samples
from a given channel. All subframes within a frame will contain
the same number of samples.
*_Blocking_*
The size used for blocking the audio data has a direct effect on the
compression ratio. If the block size is too small, the resulting large
number of frames mean that excess bits will be wasted on frame headers.
If the block size is too large, the characteristics of the signal may
vary so much that the encoder will be unable to find a good predictor.
In order to simplify encoder/decoder design, FLAC imposes a minimum
block size of 16 samples, and a maximum block size of 65535 samples.
This range covers the optimal size for all of the audio data FLAC supports.
Currently the reference encoder uses a fixed block size, optimized on
the sample rate of the input. Future versions may vary the block size
depending on the characteristics of the signal.
Blocked data is passed to the predictor stage one subblock (channel) at
a time. Each subblock is independently coded into a subframe, and the
subframes are concatenated into a frame. Because each channel is coded
separately, it means that one channel of a stereo frame may be encoded
as a constant subframe, and the other an LPC subframe.
*_Interchannel Decorrelation_*
In stereo streams, many times there is an exploitable amount of
correlation between the left and right channels. FLAC allows the frames
of stereo streams to have different channel assignments, and an encoder
may choose to use the best representation on a frame-by-frame basis.
* *Independent*. The left and right channels are coded independently.
* *Mid-side*. The left and right channels are transformed into mid
and side channels. The mid channel is the midpoint (average) of
the left and right signals, and the side is the difference signal
(left minus right).
* *Left-side*. The left channel and side channel are coded.
* *Right-side*. The right channel and side channel are coded
Surprisingly, the left-side and right-side forms can be the most
efficient in many frames, even though the raw number of bits per sample
needed for the original signal is slightly more than that needed for
independent or mid-side coding.
*_Prediction_*
FLAC uses four methods for modeling the input signal:
* *Verbatim*. This is essentially a zero-order predictor of the
signal. The predicted signal is zero, meaning the residual is the
signal itself, and the compression is zero. This is the baseline
against which the other predictors are measured. If you feed
random data to the encoder, the verbatim predictor will probably
be used for every subblock. Since the raw signal is not actually
passed through the residual coding stage (it is added to the
stream 'verbatim'), the encoding results will not be the same as a
zero-order linear predictor.
* *Constant*. This predictor is used whenever the subblock is pure
DC ("digital silence"), i.e. a constant value throughout. The
signal is run-length encoded and added to the stream.
* *Fixed linear predictor*. FLAC uses a class of
computationally-efficient fixed linear predictors (for a good
description, see audiopak
and
shorten
).
FLAC adds a fourth-order predictor to the zero-to-third-order
predictors used by Shorten. Since the predictors are fixed, the
predictor order is the only parameter that needs to be stored in
the compressed stream. The error signal is then passed to the
residual coder.
* *FIR Linear prediction*. For more accurate modeling (at a cost of
slower encoding), FLAC supports up to 32nd order FIR linear
prediction (again, for information on linear prediction, see
audiopak
and shorten
).
The reference encoder uses the Levinson-Durbin method for
calculating the LPC coefficients from the autocorrelation
coefficients, and the coefficients are quantized before computing
the residual. Whereas encoders such as Shorten used a fixed
quantization for the entire input, FLAC allows the quantized
coefficient precision to vary from subframe to subframe. The FLAC
reference encoder estimates the optimal precision to use based on
the block size and dynamic range of the original signal.
*_Residual Coding_*
FLAC currently defines two similar methods for the coding of the error
signal from the prediction stage. The error signal is coded using Rice
codes in one of two ways: 1) the encoder estimates a single Rice
parameter based on the variance of the residual and Rice codes the
entire residual using this parameter; 2) the residual is partitioned
into several equal-length regions of contiguous samples, and each region
is coded with its own Rice parameter based on the region's mean. (Note
that the first method is a special case of the second method with one
partition, except the Rice parameter is based on the residual variance
instead of the mean.)
The FLAC format has reserved space for other coding methods. Some
possiblities for volunteers would be to explore better context-modeling
of the Rice parameter, or Huffman coding. See LOCO-I
and pucrunch
for
descriptions of several universal codes.
*_Format_*
This section specifies the FLAC bitstream format. FLAC has no format
version information, but it does contain reserved space in several
places. Future versions of the format may use this reserved space safely
without breaking the format of older streams. Older decoders may choose
to abort decoding or skip data encoded with newer methods. Apart from
reserved patterns, in places the format specifies invalid patterns,
meaning that the patterns may never appear in any valid bitstream, in
any prior, present, or future versions of the format. These invalid
patterns are usually used to make the synchronization mechanism more robust.
All numbers used in a FLAC bitstream are integers; there are no
floating-point representations. All numbers are big-endian coded. All
numbers are unsigned unless otherwise specified.
Before the formal description of the stream, an overview might be helpful.
* A FLAC bitstream consists of the "fLaC" marker at the beginning of
the stream, followed by a mandatory metadata block (called the
STREAMINFO block), any number of other metadata blocks, then the
audio frames.
* FLAC supports up to 128 kinds of metadata blocks; currently the
following are defined:
o *STREAMINFO*: This block has information about the whole
stream, like sample rate, number of channels, total number
of samples, etc. It must be present as the first metadata
block in the stream. Other metadata blocks may follow, and
ones that the decoder doesn't understand, it will skip.
o *APPLICATION*: This block is for use by third-party
applications. The only mandatory field is a 32-bit
identifier. This ID is granted upon request to an
application by the FLAC maintainers. The remainder is of the
block is defined by the registered application. Visit the
registration page if you would like to register an
ID for your application with FLAC.
o *PADDING*: This block allows for an arbitrary amount of
padding. The contents of a PADDING block have no meaning.
This block is useful when it is known that metadata will be
edited after encoding; the user can instruct the encoder to
reserve a PADDING block of sufficient size so that when
metadata is added, it will simply overwrite the padding
(which is relatively quick) instead of having to insert it
into the right place in the existing file (which would
normally require rewriting the entire file).
o *SEEKTABLE*: This is an optional block for storing seek
points. It is possible to seek to any given sample in a FLAC
stream without a seek table, but the delay can be
unpredictable since the bitrate may vary widely within a
stream. By adding seek points to a stream, this delay can be
significantly reduced. Each seek point takes 18 bytes, so 1%
resolution within a stream adds less than 2k. There can be
only one SEEKTABLE in a stream, but the table can have any
number of seek points. There is also a special 'placeholder'
seekpoint which will be ignored by decoders but which can be
used to reserve space for future seek point insertion.
o *VORBIS_COMMENT*: This block is for storing a list of
human-readable name/value pairs. Values are encoded using
UTF-8. It is an implementation of the Vorbis comment
specification
(without the framing bit). This is the only officially
supported tagging mechanism in FLAC. There may be only one
VORBIS_COMMENT block in a stream. In some external
documentation, Vorbis comments are called FLAC tags to
lessen confusion.
o *CUESHEET*: This block is for storing various information
that can be used in a cue sheet. It supports track and index
points, compatible with Red Book CD digital audio discs, as
well as other CD-DA metadata such as media catalog number
and track ISRCs. The CUESHEET block is especially useful for
backing up CD-DA discs, but it can be used as a general
purpose cueing mechanism for playback.
o *PICTURE*: This block is for storing pictures associated
with the file, most commonly cover art from CDs. There may
be more than one PICTURE block in a file. The picture format
is similar to the APIC frame in ID3v2
. The PICTURE block
has a type, MIME type, and UTF-8 description like ID3v2, and
supports external linking via URL (though this is
discouraged). The differences are that there is no
uniqueness constraint on the description field, and the MIME
type is mandatory. The FLAC PICTURE block also includes the
resolution, color depth, and palette size so that the client
can search for a suitable picture without having to scan
them all.
* The audio data is composed of one or more audio frames. Each frame
consists of a frame header, which contains a sync code,
information about the frame like the block size, sample rate,
number of channels, et cetera, and an 8-bit CRC. The frame header
also contains either the sample number of the first sample in the
frame (for variable-blocksize streams), or the frame number (for
fixed-blocksize streams). This allows for fast, sample-accurate
seeking to be performed. Following the frame header are encoded
subframes, one for each channel, and finally, the frame is
zero-padded to a byte boundary. Each subframe has its own header
that specifies how the subframe is encoded.
* Since a decoder may start decoding in the middle of a stream,
there must be a method to determine the start of a frame. A 14-bit
sync code begins each frame. The sync code will not appear
anywhere else in the frame header. However, since it may appear in
the subframes, the decoder has two other ways of ensuring a
correct sync. The first is to check that the rest of the frame
header contains no invalid data. Even this is not foolproof since
valid header patterns can still occur within the subframes. The
decoder's final check is to generate an 8-bit CRC of the frame
header and compare this to the CRC stored at the end of the frame
header.
* Again, since a decoder may start decoding at an arbitrary frame in
the stream, each frame header must contain some basic information
about the stream because the decoder may not have access to the
STREAMINFO metadata block at the start of the stream. This
information includes sample rate, bits per sample, number of
channels, etc. Since the frame header is pure overhead, it has a
direct effect on the compression ratio. To keep the frame header
as small as possible, FLAC uses lookup tables for the most
commonly used values for frame parameters. For instance, the
sample rate part of the frame header is specified using 4 bits.
Eight of the bit patterns correspond to the commonly used sample
rates of 8/16/22.05/24/32/44.1/48/96 kHz. However, odd sample
rates can be specified by using one of the 'hint' bit patterns,
directing the decoder to find the exact sample rate at the end of
the frame header. The same method is used for specifying the block
size and bits per sample. In this way, the frame header size stays
small for all of the most common forms of audio data.
* Individual subframes (one for each channel) are coded separately
within a frame, and appear serially in the stream. In other words,
the encoded audio data is NOT channel-interleaved. This reduces
decoder complexity at the cost of requiring larger decode buffers.
Each subframe has its own header specifying the attributes of the
subframe, like prediction method and order, residual coding
parameters, etc. The header is followed by the encoded audio data
for that channel.
* FLAC specifies a subset of itself as the Subset format. The
purpose of this is to ensure that any streams encoded according to
the Subset are truly "streamable", meaning that a decoder that
cannot seek within the stream can still pick up in the middle of
the stream and start decoding. It also makes hardware decoder
implementations more practical by limiting the encoding parameters
such that decoder buffer sizes and other resource requirements can
be easily determined. flac generates Subset streams by default
unless the "--lax" command-line option is used. The Subset makes
the following limitations on what may be used in the stream:
o The blocksize bits in the frame header <#frame_header> must
be 0001-1110. The blocksize must be <=16384; if the sample
rate is <= 48000Hz, the blocksize must be <=4608.
o The sample rate bits in the frame header <#frame_header>
must be 0001-1110.
o The bits-per-sample bits in the frame header <#frame_header>
must be 001-111.
o If the sample rate is <= 48000Hz, the filter order in LPC
subframes <#subframe_lpc> must be less than or equal to 12,
i.e. the subframe type bits in the subframe header
<#subframe_header> may not be 101100-111111.
o The Rice partition order in a Rice-coded residual section
<#partitioned_rice> must be less than or equal to 8.
The following tables constitute a formal description of the FLAC format.
Numbers in angle brackets indicate how many bits are used for a given field.
*STREAM*
<32> "fLaC", the FLAC stream marker in ASCII, meaning byte 0 of the
stream is 0x66, followed by 0x4C 0x61 0x43
/METADATA_BLOCK/ <#metadata_block_streaminfo> This is the mandatory
STREAMINFO metadata block that has the basic properties of the stream
/METADATA_BLOCK/ <#metadata_block>* Zero or more metadata blocks
/FRAME/ <#frame>+ One or more audio frames
*METADATA_BLOCK*
/METADATA_BLOCK_HEADER/ <#metadata_block_header> A block header that
specifies the type and size of the metadata block data.
/METADATA_BLOCK_DATA/ <#metadata_block_data>
*METADATA_BLOCK_HEADER*
<1> Last-metadata-block flag: '1' if this block is the last metadata
block before the audio blocks, '0' otherwise.
<7> BLOCK_TYPE
* 0 : STREAMINFO
* 1 : PADDING
* 2 : APPLICATION
* 3 : SEEKTABLE
* 4 : VORBIS_COMMENT
* 5 : CUESHEET
* 6 : PICTURE
* 7-126 : reserved
* 127 : invalid, to avoid confusion with a frame sync code
<24> Length (in bytes) of metadata to follow (does not include the size
of the METADATA_BLOCK_HEADER)
*METADATA_BLOCK_DATA*
/METADATA_BLOCK_STREAMINFO/ <#metadata_block_streaminfo>
|| /METADATA_BLOCK_PADDING/ <#metadata_block_padding>
|| /METADATA_BLOCK_APPLICATION/ <#metadata_block_application>
|| /METADATA_BLOCK_SEEKTABLE/ <#metadata_block_seektable>
|| /METADATA_BLOCK_VORBIS_COMMENT/ <#metadata_block_vorbis_comment>
|| /METADATA_BLOCK_CUESHEET/ <#metadata_block_cuesheet>
|| /METADATA_BLOCK_PICTURE/ <#metadata_block_picture> The block data
must match the block type in the block header.
*METADATA_BLOCK_STREAMINFO*
<16> The minimum block size (in samples) used in the stream.
<16> The maximum block size (in samples) used in the stream. (Minimum
blocksize == maximum blocksize) implies a fixed-blocksize stream.
<24> The minimum frame size (in bytes) used in the stream. May be 0 to
imply the value is not known.
<24> The maximum frame size (in bytes) used in the stream. May be 0 to
imply the value is not known.
<20> Sample rate in Hz. Though 20 bits are available, the maximum
sample rate is limited by the structure of frame headers to 655350Hz.
Also, a value of 0 is invalid.
<3> (number of channels)-1. FLAC supports from 1 to 8 channels
<5> (bits per sample)-1. FLAC supports from 4 to 32 bits per sample.
Currently the reference encoder and decoders only support up to 24 bits
per sample.
<36> Total samples in stream. 'Samples' means inter-channel sample,
i.e. one second of 44.1Khz audio will have 44100 samples regardless of
the number of channels. A value of zero here means the number of total
samples is unknown.
<128> MD5 signature of the unencoded audio data. This allows the
decoder to determine if an error exists in the audio data even when the
error does not result in an invalid bitstream.
NOTES
* FLAC specifies a minimum block size of 16 and a maximum block size
of 65535, meaning the bit patterns corresponding to the numbers
0-15 in the minimum blocksize and maximum blocksize fields are
invalid.
*METADATA_BLOCK_PADDING*
n '0' bits (n must be a multiple of 8)
*METADATA_BLOCK_APPLICATION*
<32> Registered application ID. (Visit the registration page
to register an ID with FLAC.)
Application data (n must be a multiple of 8)
*METADATA_BLOCK_SEEKTABLE*
/SEEKPOINT/ <#seekpoint>+ One or more seek points.
NOTE
* The number of seek points is implied by the metadata header
'length' field, i.e. equal to length / 18.
*SEEKPOINT*
<64> Sample number of first sample in the target frame, or
0xFFFFFFFFFFFFFFFF for a placeholder point.
<64> Offset (in bytes) from the first byte of the first frame header to
the first byte of the target frame's header.
<16> Number of samples in the target frame.
NOTES
* For placeholder points, the second and third field values are
undefined.
* Seek points within a table must be sorted in ascending order by
sample number.
* Seek points within a table must be unique by sample number, with
the exception of placeholder points.
* The previous two notes imply that there may be any number of
placeholder points, but they must all occur at the end of the table.
*METADATA_BLOCK_VORBIS_COMMENT*
Also known as FLAC tags, the contents of a vorbis comment packet as
specified here (without
the framing bit). Note that the vorbis comment spec allows for on the
order of 2 ^ 64 bytes of data where as the FLAC metadata block is
limited to 2 ^ 24 bytes. Given the stated purpose of vorbis comments,
i.e. human-readable textual information, this limit is unlikely to be
restrictive. Also note that the 32-bit field lengths are little-endian
coded according to the vorbis spec, as opposed to the usual big-endian
coding of fixed-length integers in the rest of FLAC.
*METADATA_BLOCK_CUESHEET*
<128*8> Media catalog number, in ASCII printable characters 0x20-0x7e.
In general, the media catalog number may be 0 to 128 bytes long; any
unused characters should be right-padded with NUL characters. For CD-DA,
this is a thirteen digit number, followed by 115 NUL bytes.
<64> The number of lead-in samples. This field has meaning only for
CD-DA cuesheets; for other uses it should be 0. For CD-DA, the lead-in
is the TRACK 00 area where the table of contents is stored; more
precisely, it is the number of samples from the first sample of the
media to the first sample of the first index point of the first track.
According to the Red Book, the lead-in must be silence and CD grabbing
software does not usually store it; additionally, the lead-in must be at
least two seconds but may be longer. For these reasons the lead-in
length is stored here so that the absolute position of the first track
can be computed. Note that the lead-in stored here is the number of
samples up to the first index point of the first track, not necessarily
to INDEX 01 of the first track; even the first track may have INDEX 00
data.
<1> 1 if the CUESHEET corresponds to a Compact Disc, else 0.
<7+258*8> Reserved. All bits must be set to zero.
<8> The number of tracks. Must be at least 1 (because of the requisite
lead-out track). For CD-DA, this number must be no more than 100 (99
regular tracks and one lead-out track).
/CUESHEET_TRACK/ <#cuesheet_track>+ One or more tracks. A CUESHEET
block is required to have a lead-out track; it is always the last track
in the CUESHEET. For CD-DA, the lead-out track number must be 170 as
specified by the Red Book, otherwise is must be 255.
*CUESHEET_TRACK*
<64> Track offset in samples, relative to the beginning of the FLAC
audio stream. It is the offset to the first index point of the track.
(Note how this differs from CD-DA, where the track's offset in the TOC
is that of the track's INDEX 01 even if there is an INDEX 00.) For
CD-DA, the offset must be evenly divisible by 588 samples (588 samples =
44100 samples/sec * 1/75th of a sec).
<8> Track number. A track number of 0 is not allowed to avoid
conflicting with the CD-DA spec, which reserves this for the lead-in.
For CD-DA the number must be 1-99, or 170 for the lead-out; for
non-CD-DA, the track number must for 255 for the lead-out. It is not
required but encouraged to start with track 1 and increase sequentially.
Track numbers must be unique within a CUESHEET.
<12*8> Track ISRC. This is a 12-digit alphanumeric code; see here
and here
. A
value of 12 ASCII NUL characters may be used to denote absence of an ISRC.
<1> The track type: 0 for audio, 1 for non-audio. This corresponds to
the CD-DA Q-channel control bit 3.
<1> The pre-emphasis flag: 0 for no pre-emphasis, 1 for pre-emphasis.
This corresponds to the CD-DA Q-channel control bit 5; see here
.
<6+13*8> Reserved. All bits must be set to zero.
<8> The number of track index points. There must be at least one index
in every track in a CUESHEET except for the lead-out track, which must
have zero. For CD-DA, this number may be no more than 100.
/CUESHEET_TRACK_INDEX/ <#cuesheet_track_index>+ For all tracks except
the lead-out track, one or more track index points.
*CUESHEET_TRACK_INDEX*
<64> Offset in samples, relative to the track offset, of the index
point. For CD-DA, the offset must be evenly divisible by 588 samples
(588 samples = 44100 samples/sec * 1/75th of a sec). Note that the
offset is from the beginning of the track, not the beginning of the
audio data.
<8> The index point number. For CD-DA, an index number of 0 corresponds
to the track pre-gap. The first index in a track must have a number of 0
or 1, and subsequently, index numbers must increase by 1. Index numbers
must be unique within a track.
<3*8> Reserved. All bits must be set to zero.
*METADATA_BLOCK_PICTURE*
<32> The picture type according to the ID3v2 APIC frame:
* 0 - Other
* 1 - 32x32 pixels 'file icon' (PNG only)
* 2 - Other file icon
* 3 - Cover (front)
* 4 - Cover (back)
* 5 - Leaflet page
* 6 - Media (e.g. label side of CD)
* 7 - Lead artist/lead performer/soloist
* 8 - Artist/performer
* 9 - Conductor
* 10 - Band/Orchestra
* 11 - Composer
* 12 - Lyricist/text writer
* 13 - Recording Location
* 14 - During recording
* 15 - During performance
* 16 - Movie/video screen capture
* 17 - A bright coloured fish
* 18 - Illustration
* 19 - Band/artist logotype
* 20 - Publisher/Studio logotype
Others are reserved and should not be used. There may only be one each
of picture type 1 and 2 in a file.
<32> The length of the MIME type string in bytes.
The MIME type string, in printable ASCII characters 0x20-0x7e.
The MIME type may also be --> to signify that the data part is a URL of
the picture instead of the picture data itself.
<32> The length of the description string in bytes.
The description of the picture, in UTF-8.
<32> The width of the picture in pixels.
<32> The height of the picture in pixels.
<32> The color depth of the picture in bits-per-pixel.
<32> For indexed-color pictures (e.g. GIF), the number of colors used,
or 0 for non-indexed pictures.
<32> The length of the picture data in bytes.
The binary picture data.
*FRAME*
/FRAME_HEADER/ <#frame_header>
/SUBFRAME/ <#subframe>+ One SUBFRAME per channel.
> Zero-padding to byte alignment.
/FRAME_FOOTER/ <#frame_footer>
*FRAME_HEADER*
<14> Sync code '11111111111110'
<1> Reserved:
* 0 : mandatory value
* 1 : reserved for future use
<1> Blocking strategy:
* 0 : fixed-blocksize stream; frame header encodes the frame number
* 1 : variable-blocksize stream; frame header encodes the sample number
<4> Block size in inter-channel samples:
* 0000 : reserved
* 0001 : 192 samples
* 0010-0101 : 576 * (2^(n-2)) samples, i.e. 576/1152/2304/4608
* 0110 : get 8 bit (blocksize-1) from end of header
* 0111 : get 16 bit (blocksize-1) from end of header
* 1000-1111 : 256 * (2^(n-8)) samples, i.e.
256/512/1024/2048/4096/8192/16384/32768
<4> Sample rate:
* 0000 : get from STREAMINFO metadata block
* 0001 : 88.2kHz
* 0010 : 176.4kHz
* 0011 : 192kHz
* 0100 : 8kHz
* 0101 : 16kHz
* 0110 : 22.05kHz
* 0111 : 24kHz
* 1000 : 32kHz
* 1001 : 44.1kHz
* 1010 : 48kHz
* 1011 : 96kHz
* 1100 : get 8 bit sample rate (in kHz) from end of header
* 1101 : get 16 bit sample rate (in Hz) from end of header
* 1110 : get 16 bit sample rate (in tens of Hz) from end of header
* 1111 : invalid, to prevent sync-fooling string of 1s
<4> Channel assignment
* 0000-0111 : (number of independent channels)-1. Where defined, the
channel order follows SMPTE/ITU-R recommendations. The assignments
are as follows:
o 1 channel: mono
o 2 channels: left, right
o 3 channels: left, right, center
o 4 channels: left, right, back left, back right
o 5 channels: left, right, center, back/surround left,
back/surround right
o 6 channels: left, right, center, LFE, back/surround left,
back/surround right
o 7 channels: not defined
o 8 channels: not defined
* 1000 : left/side stereo: channel 0 is the left channel, channel 1
is the side(difference) channel
* 1001 : right/side stereo: channel 0 is the side(difference)
channel, channel 1 is the right channel
* 1010 : mid/side stereo: channel 0 is the mid(average) channel,
channel 1 is the side(difference) channel
* 1011-1111 : reserved
<3> Sample size in bits:
* 000 : get from STREAMINFO metadata block
* 001 : 8 bits per sample
* 010 : 12 bits per sample
* 011 : reserved
* 100 : 16 bits per sample
* 101 : 20 bits per sample
* 110 : 24 bits per sample
* 111 : reserved
<1> Reserved:
* 0 : mandatory value
* 1 : reserved for future use
> if(variable blocksize)
<8-56>:"UTF-8" coded sample number (decoded number is 36 bits)
else
<8-48>:"UTF-8" coded frame number (decoded number is 31 bits)
> if(blocksize bits == 011x)
8/16 bit (blocksize-1)
> if(sample rate bits == 11xx)
8/16 bit sample rate
<8> CRC-8 (polynomial = x^8 + x^2 + x^1 + x^0, initialized with 0) of
everything before the crc, including the sync code
NOTES
* The "blocking strategy" bit must be the same throughout the entire
stream.
* The "blocking strategy" bit determines how to calculate the sample
number of the first sample in the frame. If the bit is 0
(fixed-blocksize), the frame header encodes the frame number as
above, and the frame's starting sample number will be the frame
number times the blocksize. If it is 1 (variable-blocksize), the
frame header encodes the frame's starting sample number itself.
(In the case of a fixed-blocksize stream, only the last block may
be shorter than the stream blocksize; its starting sample number
will be calculated as the frame number times the previous frame's
blocksize, or zero if it is the first frame).
* The "UTF-8" coding used for the sample/frame number is the same
variable length code used to store compressed UCS-2, extended to
handle larger input.
*FRAME_FOOTER*
<16> CRC-16 (polynomial = x^16 + x^15 + x^2 + x^0, initialized with 0)
of everything before the crc, back to and including the frame header
sync code
*SUBFRAME*
/SUBFRAME_HEADER/ <#subframe_header>
/SUBFRAME_CONSTANT/ <#subframe_constant>
|| /SUBFRAME_FIXED/ <#subframe_fixed>
|| /SUBFRAME_LPC/ <#subframe_lpc>
|| /SUBFRAME_VERBATIM/ <#subframe_verbatim> The SUBFRAME_HEADER
specifies which one.
*SUBFRAME_HEADER*
<1> Zero bit padding, to prevent sync-fooling string of 1s
<6> Subframe type:
* 000000 : SUBFRAME_CONSTANT <#subframe_constant>
* 000001 : SUBFRAME_VERBATIM <#subframe_verbatim>
* 00001x : reserved
* 0001xx : reserved
* 001xxx : if(xxx <= 4) SUBFRAME_FIXED <#subframe_fixed>, xxx=order
; else reserved
* 01xxxx : reserved
* 1xxxxx : SUBFRAME_LPC <#subframe_lpc>, xxxxx=order-1
<1+k> 'Wasted bits-per-sample' flag:
* 0 : no wasted bits-per-sample in source subblock, k=0
* 1 : k wasted bits-per-sample in source subblock, k-1 follows,
unary coded; e.g. k=3 => 001 follows, k=7 => 0000001 follows.
*SUBFRAME_CONSTANT*
Unencoded constant value of the subblock, n = frame's bits-per-sample.
*SUBFRAME_FIXED*
Unencoded warm-up samples (n = frame's bits-per-sample * predictor
order).
/RESIDUAL/ <#residual> Encoded residual
*SUBFRAME_LPC*
Unencoded warm-up samples (n = frame's bits-per-sample * lpc order).
<4> (Quantized linear predictor coefficients' precision in bits)-1
(1111 = invalid).
<5> Quantized linear predictor coefficient shift needed in bits (NOTE:
this number is signed two's-complement).
Unencoded predictor coefficients (n = qlp coeff precision * lpc
order) (NOTE: the coefficients are signed two's-complement).
/RESIDUAL/ <#residual> Encoded residual
*SUBFRAME_VERBATIM*
Unencoded subblock; n = frame's bits-per-sample, i = frame's
blocksize.
*RESIDUAL*
<2> Residual coding method:
* 00 : partitioned Rice coding with 4-bit Rice parameter;
RESIDUAL_CODING_METHOD_PARTITIONED_RICE follows
* 01 : partitioned Rice coding with 5-bit Rice parameter;
RESIDUAL_CODING_METHOD_PARTITIONED_RICE2 follows
* 10-11 : reserved
/RESIDUAL_CODING_METHOD_PARTITIONED_RICE/ <#partitioned_rice> ||
/RESIDUAL_CODING_METHOD_PARTITIONED_RICE2/ <#partitioned_rice2>
*RESIDUAL_CODING_METHOD_PARTITIONED_RICE*
<4> Partition order.
/RICE_PARTITION/ <#rice_partition>+ There will be 2^order partitions.
*RICE_PARTITION*
<4(+5)> Encoding parameter:
* 0000-1110 : Rice parameter.
* 1111 : Escape code, meaning the partition is in unencoded binary
form using n bits per sample; n follows as a 5-bit number.
> Encoded residual. The number of samples (n) in the partition is
determined as follows:
* if the partition order is zero, n = frame's blocksize - predictor
order
* else if this is not the first partition of the subframe, n =
(frame's blocksize / (2^partition order))
* else n = (frame's blocksize / (2^partition order)) - predictor order
*RESIDUAL_CODING_METHOD_PARTITIONED_RICE2*
<4> Partition order.
/RICE2_PARTITION/ <#rice2_partition>+ There will be 2^order partitions.
*RICE2_PARTITION*
<5(+5)> Encoding parameter:
* 00000-11110 : Rice parameter.
* 11111 : Escape code, meaning the partition is in unencoded binary
form using n bits per sample; n follows as a 5-bit number.
> Encoded residual. The number of samples (n) in the partition is
determined as follows:
* if the partition order is zero, n = frame's blocksize - predictor
order
* else if this is not the first partition of the subframe, n =
(frame's blocksize / (2^partition order))
* else n = (frame's blocksize / (2^partition order)) - predictor order
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