| RFC : | rfc9639 |
| Title: | DNS Security Extensions (DNSSEC) |
| Date: | December 2024 |
| Status: | PROPOSED STANDARD |
Internet Engineering Task Force (IETF) M.Q.C. van Beurden
Request for Comments: 9639
Category: Standards Track A. Weaver
ISSN: 2070-1721 December 2024
Free Lossless Audio Codec (FLAC)
Abstract
This document defines the Free Lossless Audio Codec (FLAC) format and
its streamable subset. FLAC is designed to reduce the amount of
computer storage space needed to store digital audio signals. It
does this losslessly, i.e., it does so without losing information.
FLAC is free in the sense that its specification is open and its
reference implementation is open source. Compared to other lossless
audio coding formats, FLAC is a format with low complexity and can be
encoded and decoded with little computing resources. Decoding of
FLAC has been implemented independently for many different platforms,
and both encoding and decoding can be implemented without needing
floating-point arithmetic.
Status of This Memo
This is an Internet Standards Track document.
This document is a product of the Internet Engineering Task Force
(IETF). It represents the consensus of the IETF community. It has
received public review and has been approved for publication by the
Internet Engineering Steering Group (IESG). Further information on
Internet Standards is available in Section 2 of RFC 7841.
Information about the current status of this document, any errata,
and how to provide feedback on it may be obtained at
https://www.rfc-editor.org/info/rfc9639.
Copyright Notice
Copyright (c) 2024 IETF Trust and the persons identified as the
document authors. All rights reserved.
This document is subject to BCP 78 and the IETF Trust's Legal
Provisions Relating to IETF Documents
(https://trustee.ietf.org/license-info) in effect on the date of
publication of this document. Please review these documents
carefully, as they describe your rights and restrictions with respect
to this document. Code Components extracted from this document must
include Revised BSD License text as described in Section 4.e of the
Trust Legal Provisions and are provided without warranty as described
in the Revised BSD License.
Table of Contents
1. Introduction
2. Notation and Conventions
3. Definitions
4. Conceptual Overview
4.1. Blocking
4.2. Interchannel Decorrelation
4.3. Prediction
4.4. Residual Coding
5. Format Principles
6. Format Layout Overview
7. Streamable Subset
8. File-Level Metadata
8.1. Metadata Block Header
8.2. Streaminfo
8.3. Padding
8.4. Application
8.5. Seek Table
8.5.1. Seek Point
8.6. Vorbis Comment
8.6.1. Standard Field Names
8.6.2. Channel Mask
8.7. Cuesheet
8.7.1. Cuesheet Track
8.8. Picture
9. Frame Structure
9.1. Frame Header
9.1.1. Block Size Bits
9.1.2. Sample Rate Bits
9.1.3. Channels Bits
9.1.4. Bit Depth Bits
9.1.5. Coded Number
9.1.6. Uncommon Block Size
9.1.7. Uncommon Sample Rate
9.1.8. Frame Header CRC
9.2. Subframes
9.2.1. Subframe Header
9.2.2. Wasted Bits per Sample
9.2.3. Constant Subframe
9.2.4. Verbatim Subframe
9.2.5. Fixed Predictor Subframe
9.2.6. Linear Predictor Subframe
9.2.7. Coded Residual
9.3. Frame Footer
10. Container Mappings
10.1. Ogg Mapping
10.2. Matroska Mapping
10.3. ISO Base Media File Format (MP4) Mapping
11. Security Considerations
12. IANA Considerations
12.1. Media Type Registration
12.2. FLAC Application Metadata Block IDs Registry
13. References
13.1. Normative References
13.2. Informative References
Appendix A. Numerical Considerations
A.1. Determining the Necessary Data Type Size
A.2. Stereo Decorrelation
A.3. Prediction
A.4. Residual
A.5. Rice Coding
Appendix B. Past Format Changes
B.1. Addition of Blocking Strategy Bit
B.2. Restriction of Encoded Residual Samples
B.3. Addition of 5-Bit Rice Parameters
B.4. Restriction of LPC Shift to Non-negative Values
Appendix C. Interoperability Considerations
C.1. Features outside of the Streamable Subset
C.2. Variable Block Size
C.3. 5-Bit Rice Parameters
C.4. Rice Escape Code
C.5. Uncommon Block Size
C.6. Uncommon Bit Depth
C.7. Multi-Channel Audio and Uncommon Sample Rates
C.8. Changing Audio Properties Mid-Stream
Appendix D. Examples
D.1. Decoding Example 1
D.1.1. Example File 1 in Hexadecimal Representation
D.1.2. Example File 1 in Binary Representation
D.1.3. Signature and Streaminfo
D.1.4. Audio Frames
D.2. Decoding Example 2
D.2.1. Example File 2 in Hexadecimal Representation
D.2.2. Example File 2 in Binary Representation (Only Audio
Frames)
D.2.3. Streaminfo Metadata Block
D.2.4. Seek Table
D.2.5. Vorbis Comment
D.2.6. Padding
D.2.7. First Audio Frame
D.2.8. Second Audio Frame
D.2.9. MD5 Checksum Verification
D.3. Decoding Example 3
D.3.1. Example File 3 in Hexadecimal Representation
D.3.2. Example File 3 in Binary Representation (Only Audio
Frame)
D.3.3. Streaminfo Metadata Block
D.3.4. Audio Frame
Acknowledgments
Authors' Addresses
1. Introduction
This document defines the Free Lossless Audio Codec (FLAC) format and
its streamable subset. FLAC files and streams can code for pulse-
code modulated (PCM) audio with 1 to 8 channels, sample rates from 1
to 1048575 hertz, and bit depths from 4 to 32 bits. Most tools for
coding to and decoding from the FLAC format have been optimized for
CD-audio, which is PCM audio with 2 channels, a sample rate of 44.1
kHz, and a bit depth of 16 bits.
FLAC is able to achieve lossless compression because samples in audio
signals tend to be highly correlated with their close neighbors. In
contrast with general-purpose compressors, which often use
dictionaries, do run-length coding, or exploit long-term repetition,
FLAC removes redundancy solely in the very short term, looking back
at 32 samples at most.
The coding methods provided by the FLAC format work best on PCM audio
signals with samples that have a signed representation and are
centered around zero. Audio signals in which samples have an
unsigned representation must be transformed to a signed
representation as described in this document in order to achieve
reasonable compression. The FLAC format is not suited for
compressing audio that is not PCM.
2. Notation and Conventions
The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
"SHOULD", "SHOULD NOT", "RECOMMENDED", "NOT RECOMMENDED", "MAY", and
"OPTIONAL" in this document are to be interpreted as described in
BCP 14 [RFC2119] [RFC8174] when, and only when, they appear in all
capitals, as shown here.
Values expressed as u(n) represent an unsigned big-endian integer
using n bits. Values expressed as s(n) represent a signed big-endian
integer using n bits, signed two's complement. Where necessary, n is
expressed as an equation using * (multiplication), / (division), +
(addition), or - (subtraction). An inclusive range of the number of
bits expressed is represented with an ellipsis, such as u(m...n).
All shifts mentioned in this document are arithmetic shifts.
While the FLAC format can store digital audio as well as other
digital signals, this document uses terminology specific to digital
audio. The use of more generic terminology was deemed less clear, so
a reader interested in non-audio use of the FLAC format is expected
to make the translation from audio-specific terms to more generic
terminology.
3. Definitions
*Lossless compression*: Reducing the amount of computer storage
space needed to store data without needing to remove or
irreversibly alter any of this data in doing so. In other words,
decompressing losslessly compressed information returns exactly
the original data.
*Lossy compression*: Like lossless compression, but instead
removing, irreversibly altering, or only approximating information
for the purpose of further reducing the amount of computer storage
space needed. In other words, decompressing lossy compressed
information returns an approximation of the original data.
*Block*: A (short) section of linear PCM audio with one or more
channels.
*Subblock*: All samples within a corresponding block for one
channel. One or more subblocks form a block, and all subblocks in
a certain block contain the same number of samples.
*Frame*: A frame header, one or more subframes, and a frame footer.
It encodes the contents of a corresponding block.
*Subframe*: An encoded subblock. All subframes within a frame code
for the same number of samples. When interchannel decorrelation
is used, a subframe can correspond to either the (per-sample)
average of two subblocks or the (per-sample) difference between
two subblocks, instead of to a subblock directly; see Section 4.2.
*Interchannel samples*: A sample count that applies to all channels.
For example, one second of 44.1 kHz audio has 44100 interchannel
samples, meaning each channel has that number of samples.
*Block size*: The number of interchannel samples contained in a
block or coded in a frame.
*Bit depth* or *bits per sample*: The number of bits used to contain
each sample. This MUST be the same for all subblocks in a block
but MAY be different for different subframes in a frame because of
interchannel decorrelation. (See Section 4.2 for details on
interchannel decorrelation.)
*Predictor*: A model used to predict samples in an audio signal
based on past samples. FLAC uses such predictors to remove
redundancy in a signal in order to be able to compress it.
*Linear predictor*: A predictor using linear prediction (see
[LinearPrediction]). This is also called *linear predictive
coding (LPC)*. With a linear predictor, each prediction is a
linear combination of past samples (hence the name). A linear
predictor has a causal discrete-time finite impulse response (see
[FIR]).
*Fixed predictor*: A linear predictor in which the model parameters
are the same across all FLAC files and thus do not need to be
stored.
*Predictor order*: The number of past samples that a predictor uses.
For example, a 4th order predictor uses the 4 samples directly
preceding a certain sample to predict it. In FLAC, samples used
in a predictor are always consecutive and are always the samples
directly before the sample that is being predicted.
*Residual*: The audio signal that remains after a predictor has been
subtracted from a subblock. If the predictor has been able to
remove redundancy from the signal, the samples of the remaining
signal (the *residual samples*) will have, on average, a numerical
value closer to zero than the original signal.
*Rice code*: A variable-length code (see [VarLengthCode]). It uses
a short code for samples close to zero and a progressively longer
code for samples further away from zero. This makes use of the
observation that residual samples are often close to zero.
*Muxing*: Short for multiplexing. Combining several streams or
files into a single stream or file. In the context of this
document, muxing specifically refers to embedding a FLAC stream in
a container as described in Section 10.
4. Conceptual Overview
Similar to many other audio coders, a FLAC file is encoded following
the steps below. To decode a FLAC file, these steps are performed in
reverse order, i.e., from bottom to top.
1. *Blocking* (see Section 4.1). The input is split up into many
contiguous blocks.
2. *Interchannel Decorrelation* (see Section 4.2). In the case of
stereo streams, the FLAC format allows for transforming the left-
right signal into a mid-side signal, a left-side signal, or a
side-right signal to remove redundancy between channels.
Choosing between any of these transformations is done
independently for each block.
3. *Prediction* (see Section 4.3). To remove redundancy in a
signal, a predictor is stored for each subblock or its
transformation as formed in the previous step. A predictor
consists of a simple mathematical description that can be used,
as the name implies, to predict a certain sample from the samples
that preceded it. As this prediction is rarely exact, the error
of this prediction is passed on to the next stage. The predictor
of each subblock is completely independent from other subblocks.
Since the methods of prediction are known to both the encoder and
decoder, only the parameters of the predictor need to be included
in the compressed stream. If no usable predictor can be found
for a certain subblock, the signal is stored uncompressed, and
the next stage is skipped.
4. *Residual Coding* (see Section 4.4). As the predictor does not
describe the signal exactly, the difference between the original
signal and the predicted signal (called the error or residual
signal) is coded losslessly. If the predictor is effective, the
residual signal will require fewer bits per sample than the
original signal. FLAC uses Rice coding, a subset of Golomb
coding, with either 4-bit or 5-bit parameters to code the
residual signal.
In addition, FLAC specifies a metadata system (see Section 8) that
allows arbitrary information about the stream to be included at the
beginning of the stream.
4.1. Blocking
The block size used for audio data has a direct effect on the
compression ratio. If the block size is too small, the resulting
large number of frames means that a disproportionate number of bytes
will be spent 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,
except for the last block, and a maximum block size of 65535 samples.
The last block is allowed to be smaller than 16 samples to be able to
match the length of the encoded audio without using padding.
While the block size does not have to be constant in a FLAC file, it
is often difficult to find the optimal arrangement of block sizes for
maximum compression. Because of this, a FLAC stream has explicitly
either a constant or variable block size throughout and stores a
block number instead of a sample number to slightly improve
compression if a stream has a constant block size.
4.2. Interchannel Decorrelation
Channels are correlated in many audio files. The FLAC format can
exploit this correlation in stereo files by coding an average of all
samples in both subblocks (a mid channel) or the difference between
all samples in both subblocks (a side channel) instead of directly
coding subblocks into subframes. The following combinations are
possible:
* *Independent*. All channels are coded independently. All non-
stereo files MUST be encoded this way.
* *Mid-side*. A left and right subblock are converted to mid and
side subframes. To calculate a sample for a mid subframe, the
corresponding left and right samples are summed, and the result is
shifted right by 1 bit. To calculate a sample for a side
subframe, the corresponding right sample is subtracted from the
corresponding left sample. On decoding, all mid channel samples
have to be shifted left by 1 bit. Also, if a side channel sample
is odd, 1 has to be added to the corresponding mid channel sample
after it has been shifted left by 1 bit. To reconstruct the left
channel, the corresponding samples in the mid and side subframes
are added and the result shifted right by 1 bit. For the right
channel, the side channel has to be subtracted from the mid
channel and the result shifted right by 1 bit.
* *Left-side*. The left subblock is coded, and the left and right
subblocks are used to code a side subframe. The side subframe is
constructed in the same way as for mid-side. To decode, the right
subblock is restored by subtracting the samples in the side
subframe from the corresponding samples in the left subframe.
* *Side-right*. The left and right subblocks are used to code a side
subframe, and the right subblock is coded. The side subframe is
constructed in the same way as for mid-side. To decode, the left
subblock is restored by adding the samples in the side subframe to
the corresponding samples in the right subframe.
The side channel needs one extra bit of bit depth, as the subtraction
can produce sample values twice as large as the maximum possible in
any given bit depth. The mid channel in mid-side stereo does not
need one extra bit, as it is shifted right 1 bit. The right shift of
the mid channel does not lead to lossy behavior because an odd sample
in the mid subframe must always be accompanied by a corresponding odd
sample in the side subframe, which means the lost least-significant
bit can be restored by taking it from the sample in the side
subframe.
4.3. Prediction
The FLAC format has four methods for modeling the input signal:
1. *Verbatim*. Samples are stored directly, without any modeling.
This method is used for inputs with little correlation. Since
the raw signal is not actually passed through the residual coding
stage (it is added to the stream "verbatim"), this method is
different from using a zero-order fixed predictor.
2. *Constant*. A single sample value is stored. This method is used
whenever a signal is pure DC ("digital silence"), i.e., a
constant value throughout.
3. *Fixed predictor*. Samples are predicted with one of five fixed
(i.e., predefined) predictors, and the error of this prediction
is processed by the residual coder. These fixed predictors are
well suited for predicting simple waveforms. Since the
predictors are fixed, no predictor coefficients are stored. From
a mathematical point of view, the predictors work by
extrapolating the signal from the previous samples. The number
of previous samples used is equal to the predictor order. For
more information, see Section 9.2.5.
4. *Linear predictor*. Samples are predicted using past samples and
a set of predictor coefficients, and the error of this prediction
is processed by the residual coder. Compared to a fixed
predictor, using a generic linear predictor adds overhead as
predictor coefficients need to be stored. Therefore, this method
of prediction is best suited for predicting more complex
waveforms, where the added overhead is offset by space savings in
the residual coding stage resulting from more accurate
prediction. A linear predictor in FLAC has two parameters
besides the predictor coefficients and the predictor order: the
number of bits with which each coefficient is stored (the
coefficient precision) and a prediction right shift. A
prediction is formed by taking the sum of multiplying each
predictor coefficient with the corresponding past sample and
dividing that sum by applying the specified right shift. For
more information, see Section 9.2.6.
A FLAC encoder is free to select any of the above methods to model
the input. However, to ensure lossless coding, the following
exceptions apply:
* When the samples that need to be stored do not all have the same
value (i.e., the signal is not constant), a constant subframe
cannot be used.
* When an encoder is unable to find a fixed or linear predictor for
which all residual samples are representable in 32-bit signed
integers as stated in Section 9.2.7, a verbatim subframe is used.
For more information on fixed and linear predictors, see
[Lossless-Compression] and [Robinson-TR156].
4.4. Residual Coding
If a subframe uses a predictor to approximate the audio signal, a
residual is stored to "correct" the approximation to the exact value.
When an effective predictor is used, the average numerical value of
the residual samples is smaller than that of the samples before
prediction. While having smaller values on average, it is possible
that a few "outlier" residual samples are much larger than any of the
original samples. Sometimes these outliers even exceed the range
that the bit depth of the original audio offers.
To efficiently code such a stream of relatively small numbers with an
occasional outlier, Rice coding (a subset of Golomb coding) is used.
Depending on how small the numbers are that have to be coded, a Rice
parameter is chosen. The numerical value of each residual sample is
split into two parts by dividing it by 2^(Rice parameter), creating a
quotient and a remainder. The quotient is stored in unary form and
the remainder in binary form. If indeed most residual samples are
close to zero and a suitable Rice parameter is chosen, this form of
coding, with a so-called variable-length code, uses fewer bits than
the residual in unencoded form.
As Rice codes can only handle unsigned numbers, signed numbers are
zigzag encoded to a so-called folded residual. See Section 9.2.7 for
a more thorough explanation.
Quite often, the optimal Rice parameter varies over the course of a
subframe. To accommodate this, the residual can be split up into
partitions, where each partition has its own Rice parameter. To keep
overhead and complexity low, the number of partitions used in a
subframe is limited to powers of two.
The FLAC format uses two forms of Rice coding, which only differ in
the number of bits used for encoding the Rice parameter, either 4 or
5 bits.
5. Format Principles
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 when encountering data
that is encoded using methods they do not recognize. Apart from
reserved patterns, the format specifies forbidden patterns in certain
places, meaning that the patterns MUST NOT appear in any bitstream.
They are listed in the following table.
+=========================================+=============+
| Description | Reference |
+=========================================+=============+
| Metadata block type 127 | Section 8.1 |
+-----------------------------------------+-------------+
| Minimum and maximum block sizes smaller | Section 8.2 |
| than 16 in streaminfo metadata block | |
+-----------------------------------------+-------------+
| Sample rate bits 0b1111 | Section |
| | 9.1.2 |
+-----------------------------------------+-------------+
| Uncommon block size 65536 | Section |
| | 9.1.6 |
+-----------------------------------------+-------------+
| Predictor coefficient precision bits | Section |
| 0b1111 | 9.2.6 |
+-----------------------------------------+-------------+
| Negative predictor right shift | Section |
| | 9.2.6 |
+-----------------------------------------+-------------+
Table 1
All numbers used in a FLAC bitstream are integers; there are no
floating-point representations. All numbers are big-endian coded,
except the field lengths used in Vorbis comments (see Section 8.6),
which are little-endian coded. This exception for Vorbis comments is
to keep as much commonality as possible with Vorbis comments as used
by the Vorbis codec (see [Vorbis]). All numbers are unsigned except
linear predictor coefficients, the linear prediction shift (see
Section 9.2.6), and numbers that directly represent samples, which
are signed. None of these restrictions apply to application metadata
blocks or to Vorbis comment field contents.
All samples encoded to and decoded from the FLAC format MUST be in a
signed representation.
There are several ways to convert unsigned sample representations to
signed sample representations, but the coding methods provided by the
FLAC format work best on samples that have numerical values that are
centered around zero, i.e., have no DC offset. In most unsigned
audio formats, signals are centered around halfway within the range
of the unsigned integer type used. If that is the case, converting
sample representations by first copying the number to a signed
integer with a sufficient range and then subtracting half of the
range of the unsigned integer type results in a signal with samples
centered around 0.
Unary coding in a FLAC bitstream is done with zero bits terminated
with a one bit, e.g., the number 5 is coded unary as 0b000001. This
prevents the frame sync code from appearing in unary-coded numbers.
When a FLAC file contains data that is forbidden or otherwise not
valid, decoder behavior is left unspecified. A decoder MAY choose to
stop decoding upon encountering such data. Examples of such data
include the following:
* One or more decoded sample values exceed the range offered by the
bit depth as coded for that frame. For example, in a frame with a
bit depth of 8 bits, any samples not in the inclusive range from
-128 to 127 are not valid.
* The number of wasted bits (see Section 9.2.2) used by a subframe
is such that the bit depth of that subframe (see Section 9.2.3 for
a description of subframe bit depth) equals zero or is negative.
* A frame header Cyclic Redundancy Check (CRC) (see Section 9.1.8)
or frame footer CRC (see Section 9.3) does not validate.
* One of the forbidden bit patterns described in Table 1 is used.
6. Format Layout Overview
A FLAC bitstream consists of the fLaC (i.e., 0x664C6143) marker at
the beginning of the stream, followed by a mandatory metadata block
(called the streaminfo metadata block), any number of other metadata
blocks, and then the audio frames.
FLAC supports 127 kinds of metadata blocks; currently, 7 kinds are
defined in Section 8.
The audio data is composed of one or more audio frames. Each frame
consists of a frame header that contains a sync code, information
about the frame (like the block size, sample rate, and number of
channels), and an 8-bit CRC. The frame header also contains either
the sample number of the first sample in the frame (for variable
block size streams) or the frame number (for fixed block size
streams). This allows for fast, sample-accurate seeking to be
performed. Following the frame header are encoded subframes, one for
each channel. The frame is then zero-padded to a byte boundary and
finished with a frame footer containing a checksum for the frame.
Each subframe has its own header that specifies how the subframe is
encoded.
In order to allow a decoder to start decoding at any place in the
stream, each frame starts with a byte-aligned 15-bit sync code.
However, since it is not guaranteed that the sync code does not
appear elsewhere in the frame, the decoder can check that it synced
correctly by parsing the rest of the frame header and validating the
frame header CRC.
Furthermore, to allow a decoder to start decoding at any place in the
stream even without having received a streaminfo metadata block, each
frame header contains some basic information about the stream. This
information includes sample rate, bits per sample, number of
channels, etc. Since the frame header is 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 properties. When a certain property has a value
that is not covered by the lookup table, the decoder is directed to
find the value of that property (for example, the sample rate) at the
end of the frame header or in the streaminfo metadata block. If a
frame header refers to the streaminfo metadata block, the file is not
"streamable"; see Section 7 for details. By using lookup tables, the
file is streamable and the frame header size is small for 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. Each subframe header is followed by the encoded
audio data for that channel.
7. Streamable Subset
The FLAC format specifies a subset of itself as the FLAC streamable
subset. The purpose of this is to ensure that any streams encoded
according to this 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 in such a way that decoder buffer sizes and other resource
requirements can be easily determined. The streamable subset makes
the following limitations on what MAY be used in the stream:
* The sample rate bits (see Section 9.1.2) in the frame header MUST
be 0b0001-0b1110, i.e., the frame header MUST NOT refer to the
streaminfo metadata block to describe the sample rate.
* The bit depth bits (see Section 9.1.4) in the frame header MUST be
0b001-0b111, i.e., the frame header MUST NOT refer to the
streaminfo metadata block to describe the bit depth.
* The stream MUST NOT contain blocks with more than 16384
interchannel samples, i.e., the maximum block size must not be
larger than 16384.
* Audio with a sample rate less than or equal to 48000 Hz MUST NOT
be contained in blocks with more than 4608 interchannel samples,
i.e., the maximum block size used for this audio must not be
larger than 4608.
* Linear prediction subframes (see Section 9.2.6) containing audio
with a sample rate less than or equal to 48000 Hz MUST have a
predictor order less than or equal to 12, i.e., the subframe type
bits in the subframe header (see Section 9.2.1) MUST NOT be
0b101100-0b111111.
* The Rice partition order (see Section 9.2.7) MUST be less than or
equal to 8.
* The channel ordering MUST be equal to one defined in
Section 9.1.3, i.e., the FLAC file MUST NOT need a
WAVEFORMATEXTENSIBLE_CHANNEL_MASK tag to describe the channel
ordering. See Section 8.6.2 for details.
8. File-Level Metadata
At the start of a FLAC file or stream, following the fLaC ASCII file
signature, one or more metadata blocks MUST be present before any
audio frames appear. The first metadata block MUST be a streaminfo
metadata block.
8.1. Metadata Block Header
Each metadata block starts with a 4-byte header. The first bit in
this header flags whether a metadata block is the last one. It is 0
when other metadata blocks follow; otherwise, it is 1. The 7
remaining bits of the first header byte contain the type of the
metadata block as an unsigned number between 0 and 126, according to
the following table. A value of 127 (i.e., 0b1111111) is forbidden.
The three bytes that follow code for the size of the metadata block
in bytes, excluding the 4 header bytes, as an unsigned number coded
big-endian.
+=========+=======================================================+
| Value | Metadata Block Type |
+=========+=======================================================+
| 0 | Streaminfo |
+---------+-------------------------------------------------------+
| 1 | Padding |
+---------+-------------------------------------------------------+
| 2 | Application |
+---------+-------------------------------------------------------+
| 3 | Seek table |
+---------+-------------------------------------------------------+
| 4 | Vorbis comment |
+---------+-------------------------------------------------------+
| 5 | Cuesheet |
+---------+-------------------------------------------------------+
| 6 | Picture |
+---------+-------------------------------------------------------+
| 7 - 126 | Reserved |
+---------+-------------------------------------------------------+
| 127 | Forbidden (to avoid confusion with a frame sync code) |
+---------+-------------------------------------------------------+
Table 2
8.2. Streaminfo
The streaminfo metadata block has information about the whole stream,
such as 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. There MUST be no more than one
streaminfo metadata block per FLAC stream.
If the streaminfo metadata block contains incorrect or incomplete
information, decoder behavior is left unspecified (i.e., it is up to
the decoder implementation). A decoder MAY choose to stop further
decoding when the information supplied by the streaminfo metadata
block turns out to be incorrect or contains forbidden values. A
decoder accepting information from the streaminfo metadata block
(most significantly, the maximum frame size, maximum block size,
number of audio channels, number of bits per sample, and total number
of samples) without doing further checks during decoding of audio
frames could be vulnerable to buffer overflows. See also Section 11.
The following table describes the streaminfo metadata block in order,
excluding the metadata block header.
+========+=================================================+
| Data | Description |
+========+=================================================+
| u(16) | The minimum block size (in samples) used in the |
| | stream, excluding the last block. |
+--------+-------------------------------------------------+
| u(16) | The maximum block size (in samples) used in the |
| | stream. |
+--------+-------------------------------------------------+
| u(24) | The minimum frame size (in bytes) used in the |
| | stream. A value of 0 signifies that the value |
| | is not known. |
+--------+-------------------------------------------------+
| u(24) | The maximum frame size (in bytes) used in the |
| | stream. A value of 0 signifies that the value |
| | is not known. |
+--------+-------------------------------------------------+
| u(20) | Sample rate in Hz. |
+--------+-------------------------------------------------+
| u(3) | (number of channels)-1. FLAC supports from 1 |
| | to 8 channels. |
+--------+-------------------------------------------------+
| u(5) | (bits per sample)-1. FLAC supports from 4 to |
| | 32 bits per sample. |
+--------+-------------------------------------------------+
| u(36) | Total number of interchannel samples in the |
| | stream. A value of 0 here means the number of |
| | total samples is unknown. |
+--------+-------------------------------------------------+
| u(128) | MD5 checksum of the unencoded audio data. This |
| | allows the decoder to determine if an error |
| | exists in the audio data even when, despite the |
| | error, the bitstream itself is valid. A value |
| | of 0 signifies that the value is not known. |
+--------+-------------------------------------------------+
Table 3
The minimum block size and the maximum block size MUST be in the
16-65535 range. The minimum block size MUST be equal to or less than
the maximum block size.
Any frame but the last one MUST have a block size equal to or greater
than the minimum block size and MUST have a block size equal to or
less than the maximum block size. The last frame MUST have a block
size equal to or less than the maximum block size; it does not have
to comply to the minimum block size because the block size of that
frame must be able to accommodate the length of the audio data the
stream contains.
If the minimum block size is equal to the maximum block size, the
file contains a fixed block size stream, as the minimum block size
excludes the last block. Note that in the case of a stream with a
variable block size, the actual maximum block size MAY be smaller
than the maximum block size listed in the streaminfo metadata block,
and the actual smallest block size excluding the last block MAY be
larger than the minimum block size listed in the streaminfo metadata
block. This is because the encoder has to write these fields before
receiving any input audio data and cannot know beforehand what block
sizes it will use, only between what bounds the block sizes will be
chosen.
The sample rate MUST NOT be 0 when the FLAC file contains audio. A
sample rate of 0 MAY be used when non-audio is represented. This is
useful if data is encoded that is not along a time axis or when the
sample rate of the data lies outside the range that FLAC can
represent in the streaminfo metadata block. If a sample rate of 0 is
used, it is recommended to store the meaning of the encoded content
in a Vorbis comment field (see Section 8.6) or an application
metadata block (see Section 8.4). This document does not define such
metadata.
The MD5 checksum is computed by applying the MD5 message-digest
algorithm in [RFC1321]. The message to this algorithm consists of
all the samples of all channels interleaved, represented in signed,
little-endian form. This interleaving is on a per-sample basis, so
for a stereo file, this means the first sample of the first channel,
then the first sample of the second channel, then the second sample
of the first channel, etc. Before computing the checksum, all
samples must be byte-aligned. If the bit depth is not a whole number
of bytes, the value of each sample is sign-extended to the next whole
number of bytes.
In the case of a 2-channel stream with 6-bit samples, bits will be
lined up as follows:
SSAAAAAASSBBBBBBSSCCCCCC
^ ^ ^ ^ ^ ^
| | | | | Bits of 2nd sample of 1st channel
| | | | Sign extension bits of 2nd sample of 2nd channel
| | | Bits of 1st sample of 2nd channel
| | Sign extension bits of 1st sample of 2nd channel
| Bits of 1st sample of 1st channel
Sign extension bits of 1st sample of 1st channel
In the case of a 1-channel stream with 12-bit samples, bits are lined
up in little-endian byte order as follows:
AAAAAAAASSSSAAAABBBBBBBBSSSSBBBB
^ ^ ^ ^ ^ ^
| | | | | Most-significant 4 bits of 2nd sample
| | | | Sign extension bits of 2nd sample
| | | Least-significant 8 bits of 2nd sample
| | Most-significant 4 bits of 1st sample
| Sign extension bits of 1st sample
Least-significant 8 bits of 1st sample
8.3. Padding
The padding metadata block allows for an arbitrary amount of padding.
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 existing file (which would normally
require rewriting the entire file). There MAY be one or more padding
metadata blocks per FLAC stream.
+======+======================================================+
| Data | Description |
+======+======================================================+
| u(n) | n "0" bits (n MUST be a multiple of 8, i.e., a whole |
| | number of bytes, and MAY be zero). n is 8 times the |
| | size described in the metadata block header. |
+------+------------------------------------------------------+
Table 4
8.4. Application
The application metadata block is for use by third-party
applications. The only mandatory field is a 32-bit application
identifier (application ID). Application IDs are registered in the
IANA "FLAC Application Metadata Block IDs" registry (see
Section 12.2).
+=======+===================================================+
| Data | Description |
+=======+===================================================+
| u(32) | Registered application ID. |
+-------+---------------------------------------------------+
| u(n) | Application data (n MUST be a multiple of 8, |
| | i.e., a whole number of bytes). n is 8 times the |
| | size described in the metadata block header minus |
| | the 32 bits already used for the application ID. |
+-------+---------------------------------------------------+
Table 5
8.5. Seek Table
The seek table metadata block can be used to store 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. There MUST NOT be more than one
seek table metadata block in a stream, but the table can have any
number of seek points.
Each seek point takes 18 bytes, so a seek table with 1% resolution
within a stream adds less than 2 kilobytes of data. The number of
seek points is implied by the size described in the metadata block
header, i.e., equal to size / 18. There is also a special
"placeholder" seek point that will be ignored by decoders but can be
used to reserve space for future seek point insertion.
+=============+=============================+
| Data | Description |
+=============+=============================+
| Seek points | Zero or more seek points as |
| | defined in Section 8.5.1. |
+-------------+-----------------------------+
Table 6
A seek table is generally not usable for seeking in a FLAC file
embedded in a container (see Section 10), as such containers usually
interleave FLAC data with other data and the offsets used in seek
points are those of an unmuxed FLAC stream. Also, containers often
provide their own seeking methods. However, it is possible to store
the seek table in the container along with other metadata when muxing
a FLAC file, so this stored seek table can be restored when demuxing
the FLAC stream into a standalone FLAC file.
8.5.1. Seek Point
+=======+==========================================================+
| Data | Description |
+=======+==========================================================+
| u(64) | Sample number of the first sample in the target frame or |
| | 0xFFFFFFFFFFFFFFFF for a placeholder point. |
+-------+----------------------------------------------------------+
| u(64) | Offset (in bytes) from the first byte of the first frame |
| | header to the first byte of the target frame's header. |
+-------+----------------------------------------------------------+
| u(16) | Number of samples in the target frame. |
+-------+----------------------------------------------------------+
Table 7
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.
* The sample offsets are those of an unmuxed FLAC stream. The
offsets MUST NOT be updated on muxing to reflect the new offsets
of FLAC frames in a container.
8.6. Vorbis Comment
A Vorbis comment metadata block contains human-readable information
coded in UTF-8. The name "Vorbis comment" points to the fact that
the Vorbis codec stores such metadata in almost the same way (see
[Vorbis]). A Vorbis comment metadata block consists of a vendor
string optionally followed by a number of fields, which are pairs of
field names and field contents. The vendor string contains the name
of the program that generated the file or stream. The fields contain
metadata describing various aspects of the contained audio. Many
users refer to these fields as "FLAC tags" or simply as "tags". A
FLAC file MUST NOT contain more than one Vorbis comment metadata
block.
In a Vorbis comment metadata block, the metadata block header is
directly followed by 4 bytes containing the length in bytes of the
vendor string as an unsigned number coded little-endian. The vendor
string follows, is UTF-8 coded and is not terminated in any way.
Following the vendor string are 4 bytes containing the number of
fields that are in the Vorbis comment block, stored as an unsigned
number coded little-endian. If this number is non-zero, it is
followed by the fields themselves, each of which is stored with a
4-byte length. For each field, the field length in bytes is stored
as a 4-byte unsigned number coded little-endian. The field itself
follows it. Like the vendor string, the field is UTF-8 coded and not
terminated in any way.
Each field consists of a field name and field contents, separated by
an = character. The field name MUST only consist of UTF-8 code
points U+0020 through U+007E, excluding U+003D, which is the =
character. In other words, the field name can contain all printable
ASCII characters except the equals sign. The evaluation of the field
names MUST be case insensitive, so U+0041 through 0+005A (A-Z) MUST
be considered equivalent to U+0061 through U+007A (a-z). The field
contents can contain any UTF-8 character.
Note that the Vorbis comment as used in Vorbis allows for 2^64 bytes
of data whereas the FLAC metadata block is limited to 2^24 bytes.
Given the stated purpose of Vorbis comments, i.e., human-readable
textual information, the FLAC metadata block limit is unlikely to be
restrictive. Also, note that the 32-bit field lengths are coded
little-endian as opposed to the usual big-endian coding of fixed-
length integers in the rest of the FLAC format.
8.6.1. Standard Field Names
Only one standard field name is defined: the channel mask field (see
Section 8.6.2). No other field names are defined because the
applicability of any field name is strongly tied to the content it is
associated with. For example, field names that are useful for
describing files that contain a single work of music would be
unusable when labeling archived broadcasts, recordings of any kind,
or a collection of music works. Even when describing a single work
of music, different conventions exist depending on the kind of music:
orchestral music differs from music by solo artists or bands.
Despite the fact that no field names are formally defined, there is a
general trend among devices and software capable of FLAC playback
that are meant to play music. Most of those recognize at least the
following field names:
Title: Name of the current work.
Artist: Name of the artist generally responsible for the current
work. For orchestral works, this is usually the composer;
otherwise, it is often the performer.
Album: Name of the collection the current work belongs to.
For a more comprehensive list of possible field names suited for
describing a single work of music in various genres, the list of tags
used in the MusicBrainz project is suggested; see [MusicBrainz].
8.6.2. Channel Mask
Besides fields containing information about the work itself, one
field is defined for technical reasons:
WAVEFORMATEXTENSIBLE_CHANNEL_MASK. This field is used to communicate
that the channels in a file differ from the default channels defined
in Section 9.1.3. For example, by default, a FLAC file containing
two channels is interpreted to contain a left and right channel, but
with this field, it is possible to describe different channel
contents.
The channel mask consists of flag bits indicating which channels are
present. The flags only signal which channels are present, not in
which order, so if a file to be encoded has channels that are ordered
differently, they have to be reordered. This mask is stored with a
hexadecimal representation preceded by 0x; see the examples below.
Please note that a file in which the channel order is defined through
the WAVEFORMATEXTENSIBLE_CHANNEL_MASK is not streamable (see
Section 7), as the field is not found in each frame header. The mask
bits can be found in the following table.
+============+=============================+
| Bit Number | Channel Description |
+============+=============================+
| 0 | Front left |
+------------+-----------------------------+
| 1 | Front right |
+------------+-----------------------------+
| 2 | Front center |
+------------+-----------------------------+
| 3 | Low-frequency effects (LFE) |
+------------+-----------------------------+
| 4 | Back left |
+------------+-----------------------------+
| 5 | Back right |
+------------+-----------------------------+
| 6 | Front left of center |
+------------+-----------------------------+
| 7 | Front right of center |
+------------+-----------------------------+
| 8 | Back center |
+------------+-----------------------------+
| 9 | Side left |
+------------+-----------------------------+
| 10 | Side right |
+------------+-----------------------------+
| 11 | Top center |
+------------+-----------------------------+
| 12 | Top front left |
+------------+-----------------------------+
| 13 | Top front center |
+------------+-----------------------------+
| 14 | Top front right |
+------------+-----------------------------+
| 15 | Top rear left |
+------------+-----------------------------+
| 16 | Top rear center |
+------------+-----------------------------+
| 17 | Top rear right |
+------------+-----------------------------+
Table 8
Following are three examples:
* A file has a single channel -- an LFE channel. The Vorbis comment
field is WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x8.
* A file has four channels -- front left, front right, top front
left, and top front right. The Vorbis comment field is
WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x5003.
* An input has four channels -- back center, top front center, front
center, and top rear center in that order. These have to be
reordered to front center, back center, top front center, and top
rear center. The Vorbis comment field added is
WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x12104.
WAVEFORMATEXTENSIBLE_CHANNEL_MASK fields MAY be padded with zeros,
for example, 0x0008 for a single LFE channel. Parsing of
WAVEFORMATEXTENSIBLE_CHANNEL_MASK fields MUST be case-insensitive for
both the field name and the field contents.
A WAVEFORMATEXTENSIBLE_CHANNEL_MASK field of 0x0 can be used to
indicate that none of the audio channels of a file correlate with
speaker positions. This is the case when audio needs to be decoded
into speaker positions (e.g., Ambisonics B-format audio) or when a
multitrack recording is contained.
It is possible for a WAVEFORMATEXTENSIBLE_CHANNEL_MASK field to code
for fewer channels than are present in the audio. If that is the
case, the remaining channels SHOULD NOT be rendered by a playback
application unfamiliar with their purpose. For example, the
Ambisonics UHJ format is compatible with stereo playback: its first
two channels can be played back on stereo equipment, but all four
channels together can be decoded into surround sound. For that
example, the Vorbis comment field
WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x3 would be set, indicating that
the first two channels are front left and front right and other
channels do not correlate with speaker positions directly.
If audio channels not assigned to any speaker are contained and
decoding to speaker positions is possible, it is recommended to
provide metadata on how this decoding should take place in another
Vorbis comment field or an application metadata block. This document
does not define such metadata.
8.7. Cuesheet
A cuesheet metadata block can be used either to store the track and
index point structure of a Compact Disc Digital Audio (CD-DA) along
with its audio or to provide a mechanism to store locations of
interest within a FLAC file. Certain aspects of this metadata block
come directly from the CD-DA specification (called Red Book), which
is standardized as [IEC.60908.1999]. The description below is
complete, and further reference to [IEC.60908.1999] is not needed to
implement this metadata block.
The structure of a cuesheet metadata block is enumerated in the
following table.
+============+======================================================+
| Data | Description |
+============+======================================================+
| u(128*8) | Media catalog number in ASCII |
| | printable characters 0x20-0x7E. |
+------------+------------------------------------------------------+
| u(64) | Number of lead-in samples. |
+------------+------------------------------------------------------+
| u(1) | 1 if the cuesheet corresponds to a |
| | CD-DA; else 0. |
+------------+------------------------------------------------------+
| u(7+258*8) | Reserved. All bits MUST be set to |
| | zero. |
+------------+------------------------------------------------------+
| u(8) | Number of tracks in this cuesheet. |
+------------+------------------------------------------------------+
| Cuesheet | A number of structures as specified |
| tracks | in Section 8.7.1 equal to the number |
| | of tracks specified previously. |
+------------+------------------------------------------------------+
Table 9
If the media catalog number is less than 128 bytes long, it is right-
padded with 0x00 bytes. For CD-DA, this is a 13-digit number
followed by 115 0x00 bytes.
The number of lead-in samples 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
[IEC.60908.1999], the lead-in MUST be silent, 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.
The number of tracks MUST be at least 1, as a cuesheet block MUST
have a lead-out track. For CD-DA, this number MUST be no more than
100 (99 regular tracks and one lead-out track). The lead-out track
is always the last track in the cuesheet. For CD-DA, the lead-out
track number MUST be 170 as specified by [IEC.60908.1999]; otherwise,
it MUST be 255.
8.7.1. Cuesheet Track
+=============+=====================================================+
| Data | Description |
+=============+=====================================================+
| u(64) | Track offset of the first index point in |
| | samples, relative to the beginning of the |
| | FLAC audio stream. |
+-------------+-----------------------------------------------------+
| u(8) | Track number. |
+-------------+-----------------------------------------------------+
| u(12*8) | Track ISRC. |
+-------------+-----------------------------------------------------+
| u(1) | The track type: 0 for audio, 1 for non-audio. |
| | This corresponds to the CD-DA Q-channel |
| | control bit 3. |
+-------------+-----------------------------------------------------+
| u(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. |
+-------------+-----------------------------------------------------+
| u(6+13*8) | Reserved. All bits MUST be set to zero. |
+-------------+-----------------------------------------------------+
| u(8) | The number of track index points. |
+-------------+-----------------------------------------------------+
| Cuesheet | For all tracks except the lead-out track, a |
| track index | number of structures as specified in |
| points | Section 8.7.1.1 equal to the number of index |
| | points specified previously. |
+-------------+-----------------------------------------------------+
Table 10
Note that the track offset differs from the one in CD-DA, where the
track's offset in the table of contents (TOC) is that of the track's
INDEX 01 even if there is an INDEX 00. For CD-DA, the track offset
MUST be evenly divisible by 588 samples (588 samples = 44100 samples/
s * 1/75 s).
A track number of 0 is not allowed because the CD-DA specification
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 be 255 for
the lead-out. It is recommended to start with track 1 and increase
sequentially. Track numbers MUST be unique within a cuesheet.
The track ISRC (International Standard Recording Code) is a 12-digit
alphanumeric code; see [ISRC-handbook]. A value of 12 ASCII 0x00
characters MAY be used to denote the absence of an ISRC.
There MUST be at least one index point in every track in a cuesheet
except for the lead-out track, which MUST have zero. For CD-DA, the
number of index points MUST NOT be more than 100.
8.7.1.1. Cuesheet Track Index Point
+========+====================================+
| Data | Description |
+========+====================================+
| u(64) | Offset in samples, relative to the |
| | track offset, of the index point. |
+--------+------------------------------------+
| u(8) | The track index point number. |
+--------+------------------------------------+
| u(3*8) | Reserved. All bits MUST be set to |
| | zero. |
+--------+------------------------------------+
Table 11
For CD-DA, the track index point offset MUST be evenly divisible by
588 samples (588 samples = 44100 samples/s * 1/75 s). Note that the
offset is from the beginning of the track, not the beginning of the
audio data.
For CD-DA, a track index point number of 0 corresponds to the track
pre-gap. The first index point in a track MUST have a number of 0 or
1, and subsequently, index point numbers MUST increase by 1. Index
point numbers MUST be unique within a track.
8.8. Picture
The picture metadata block contains image data of a picture in some
way belonging to the audio contained in the FLAC file. Its format is
derived from the Attached Picture (APIC) frame in the ID3v2
specification; see [ID3v2]. However, contrary to the APIC frame in
ID3v2, the media type and description are prepended with a 4-byte
length field instead of being 0x00 delimited strings. A FLAC file
MAY contain one or more picture metadata blocks.
Note that while the length fields for media type, description, and
picture data are 4 bytes in length and could code for a size up to 4
GiB in theory, the total metadata block size cannot exceed what can
be described by the metadata block header, i.e., 16 MiB.
Instead of picture data, the picture metadata block can also contain
a URI as described in [RFC3986].
The structure of a picture metadata block is enumerated in the
following table.
+========+==========================================================+
| Data | Description |
+========+==========================================================+
| u(32) | The picture type according to Table 13. |
+--------+----------------------------------------------------------+
| u(32) | The length of the media type string in bytes. |
+--------+----------------------------------------------------------+
| u(n*8) | The media type string as specified by [RFC2046], |
| | or the text string --> to signify that the data |
| | part is a URI of the picture instead of the |
| | picture data itself. This field must be in |
| | printable ASCII characters 0x20-0x7E. |
+--------+----------------------------------------------------------+
| u(32) | The length of the description string in bytes. |
+--------+----------------------------------------------------------+
| u(n*8) | The description of the picture in UTF-8. |
+--------+----------------------------------------------------------+
| u(32) | The width of the picture in pixels. |
+--------+----------------------------------------------------------+
| u(32) | The height of the picture in pixels. |
+--------+----------------------------------------------------------+
| u(32) | The color depth of the picture in bits per |
| | pixel. |
+--------+----------------------------------------------------------+
| u(32) | For indexed-color pictures (e.g., GIF), the |
| | number of colors used; 0 for non-indexed |
| | pictures. |
+--------+----------------------------------------------------------+
| u(32) | The length of the picture data in bytes. |
+--------+----------------------------------------------------------+
| u(n*8) | The binary picture data. |
+--------+----------------------------------------------------------+
Table 12
The height, width, color depth, and "number of colors" fields are for
informational purposes only. Applications MUST NOT use them in
decoding the picture or deciding how to display it, but applications
MAY use them to decide whether or not to process a block (e.g., when
selecting between different picture blocks) and MAY show them to the
user. If a picture has no concept for any of these fields (e.g.,
vector images may not have a height or width in pixels) or the
content of any field is unknown, the affected fields MUST be set to
zero.
The following table contains all the defined picture types. Values
other than those listed in the table are reserved. There MAY only be
one each of picture types 1 and 2 in a file. In general practice,
many FLAC playback devices and software display the contents of a
picture metadata block, if present, with picture type 3 (front cover)
during playback.
+=======+=================================================+
| Value | Picture Type |
+=======+=================================================+
| 0 | Other |
+-------+-------------------------------------------------+
| 1 | PNG file icon of 32x32 pixels (see [RFC2083]) |
+-------+-------------------------------------------------+
| 2 | General file icon |
+-------+-------------------------------------------------+
| 3 | Front cover |
+-------+-------------------------------------------------+
| 4 | Back cover |
+-------+-------------------------------------------------+
| 5 | Liner notes page |
+-------+-------------------------------------------------+
| 6 | Media label (e.g., CD, Vinyl or Cassette label) |
+-------+-------------------------------------------------+
| 7 | Lead artist, lead performer, or soloist |
+-------+-------------------------------------------------+
| 8 | Artist or performer |
+-------+-------------------------------------------------+
| 9 | Conductor |
+-------+-------------------------------------------------+
| 10 | Band or orchestra |
+-------+-------------------------------------------------+
| 11 | Composer |
+-------+-------------------------------------------------+
| 12 | Lyricist or text writer |
+-------+-------------------------------------------------+
| 13 | Recording location |
+-------+-------------------------------------------------+
| 14 | During recording |
+-------+-------------------------------------------------+
| 15 | During performance |
+-------+-------------------------------------------------+
| 16 | Movie or video screen capture |
+-------+-------------------------------------------------+
| 17 | A bright colored fish |
+-------+-------------------------------------------------+
| 18 | Illustration |
+-------+-------------------------------------------------+
| 19 | Band or artist logotype |
+-------+-------------------------------------------------+
| 20 | Publisher or studio logotype |
+-------+-------------------------------------------------+
Table 13
The origin and use of value 17 ("A bright colored fish") is unclear.
This was copied to maintain compatibility with ID3v2. Applications
are discouraged from offering this value to users when embedding a
picture.
If a URI (not a picture) is contained in this block, the following
points apply:
* The URI can be in either absolute or relative form. If a URI is
in relative form, it is related to the URI of the FLAC content
processed.
* Applications MUST obtain explicit user approval to retrieve images
via remote protocols and to retrieve local images that are not
located in the same directory as the FLAC file being processed.
* Applications supporting linked images MUST handle unavailability
of URIs gracefully. They MAY report unavailability to the user.
* Applications MAY reject processing URIs for any reason,
particularly for security or privacy reasons.
9. Frame Structure
One or more frames follow directly after the last metadata block.
Each frame consists of a frame header, one or more subframes, padding
zero bits to achieve byte alignment, and a frame footer. The number
of subframes in each frame is equal to the number of audio channels.
Each frame header stores the audio sample rate, number of bits per
sample, and number of channels independently of the streaminfo
metadata block and other frame headers. This was done to permit
multicasting of FLAC files, but it also allows these properties to
change mid-stream. Because not all environments in which FLAC
decoders are used are able to cope with changes to these properties
during playback, a decoder MAY choose to stop decoding on such a
change. A decoder that does not check for such a change could be
vulnerable to buffer overflows. See also Section 11.
Note that storing audio with changing audio properties in FLAC
results in various practical problems. For example, these changes of
audio properties must happen on a frame boundary or the process will
not be lossless. When a variable block size is chosen to accommodate
this, note that blocks smaller than 16 samples are not allowed;
therefore, it is not possible to store an audio stream in which these
properties change within 16 samples of the last change or the start
of the file. Also, since the streaminfo metadata block can only
accommodate a single set of properties, it is only valid for part of
such an audio stream. Instead, it is RECOMMENDED to store an audio
stream with changing properties in FLAC encapsulated in a container
capable of handling such changes, as these do not suffer from the
mentioned limitations. See Section 10 for details.
9.1. Frame Header
Each frame MUST start on a byte boundary and start with the 15-bit
frame sync code 0b111111111111100. Following the sync code is the
blocking strategy bit, which MUST NOT change during the audio stream.
The blocking strategy bit is 0 for a fixed block size stream or 1 for
a variable block size stream. If the blocking strategy is known, a
decoder can include this bit when searching for the start of a frame
to reduce the possibility of encountering a false positive, as the
first two bytes of a frame are either 0xFFF8 for a fixed block size
stream or 0xFFF9 for a variable block size stream.
9.1.1. Block Size Bits
Following the frame sync code and blocking strategy bit are 4 bits
(the first 4 bits of the third byte of each frame) referred to as the
block size bits. Their value relates to the block size according to
the following table, where v is the value of the 4 bits as an
unsigned number. If the block size bits code for an uncommon block
size, this is stored after the coded number; see Section 9.1.6.
+=================+=============================================+
| Value | Block Size |
+=================+=============================================+
| 0b0000 | Reserved |
+-----------------+---------------------------------------------+
| 0b0001 | 192 |
+-----------------+---------------------------------------------+
| 0b0010 - 0b0101 | 144 * (2^v), i.e., 576, 1152, 2304, or 4608 |
+-----------------+---------------------------------------------+
| 0b0110 | Uncommon block size minus 1, stored as an |
| | 8-bit number |
+-----------------+---------------------------------------------+
| 0b0111 | Uncommon block size minus 1, stored as a |
| | 16-bit number |
+-----------------+---------------------------------------------+
| 0b1000 - 0b1111 | 2^v, i.e., 256, 512, 1024, 2048, 4096, |
| | 8192, 16384, or 32768 |
+-----------------+---------------------------------------------+
Table 14
9.1.2. Sample Rate Bits
The next 4 bits (the last 4 bits of the third byte of each frame),
referred to as the sample rate bits, contain the sample rate of the
audio according to the following table. If the sample rate bits code
for an uncommon sample rate, this is stored after the uncommon block
size; if no uncommon block size was used, this is stored after the
coded number. See Section 9.1.7.
+========+==========================================================+
| Value | Sample Rate |
+========+==========================================================+
| 0b0000 | Sample rate only stored in the |
| | streaminfo metadata block |
+--------+----------------------------------------------------------+
| 0b0001 | 88.2 kHz |
+--------+----------------------------------------------------------+
| 0b0010 | 176.4 kHz |
+--------+----------------------------------------------------------+
| 0b0011 | 192 kHz |
+--------+----------------------------------------------------------+
| 0b0100 | 8 kHz |
+--------+----------------------------------------------------------+
| 0b0101 | 16 kHz |
+--------+----------------------------------------------------------+
| 0b0110 | 22.05 kHz |
+--------+----------------------------------------------------------+
| 0b0111 | 24 kHz |
+--------+----------------------------------------------------------+
| 0b1000 | 32 kHz |
+--------+----------------------------------------------------------+
| 0b1001 | 44.1 kHz |
+--------+----------------------------------------------------------+
| 0b1010 | 48 kHz |
+--------+----------------------------------------------------------+
| 0b1011 | 96 kHz |
+--------+----------------------------------------------------------+
| 0b1100 | Uncommon sample rate in kHz, |
| | stored as an 8-bit number |
+--------+----------------------------------------------------------+
| 0b1101 | Uncommon sample rate in Hz, stored |
| | as a 16-bit number |
+--------+----------------------------------------------------------+
| 0b1110 | Uncommon sample rate in Hz divided |
| | by 10, stored as a 16-bit number |
+--------+----------------------------------------------------------+
| 0b1111 | Forbidden |
+--------+----------------------------------------------------------+
Table 15
9.1.3. Channels Bits
The next 4 bits (the first 4 bits of the fourth byte of each frame),
referred to as the channels bits, contain both the number of channels
of the audio as well as any stereo decorrelation used according to
the following table.
If a channel layout different than the ones listed in the following
table is used, this can be signaled with a
WAVEFORMATEXTENSIBLE_CHANNEL_MASK tag in a Vorbis comment metadata
block; see Section 8.6.2 for details. Note that even when such a
different channel layout is specified with a
WAVEFORMATEXTENSIBLE_CHANNEL_MASK and the channel ordering in the
following table is overridden, the channels bits still contain the
actual number of channels coded in the frame. For details on the way
left-side, side-right, and mid-side stereo are coded, see
Section 4.2.
+==========+====================================================+
| Value | Channels |
+==========+====================================================+
| 0b0000 | 1 channel: mono |
+----------+----------------------------------------------------+
| 0b0001 | 2 channels: left, right |
+----------+----------------------------------------------------+
| 0b0010 | 3 channels: left, right, center |
+----------+----------------------------------------------------+
| 0b0011 | 4 channels: front left, front right, back left, |
| | back right |
+----------+----------------------------------------------------+
| 0b0100 | 5 channels: front left, front right, front center, |
| | back/surround left, back/surround right |
+----------+----------------------------------------------------+
| 0b0101 | 6 channels: front left, front right, front center, |
| | LFE, back/surround left, back/surround right |
+----------+----------------------------------------------------+
| 0b0110 | 7 channels: front left, front right, front center, |
| | LFE, back center, side left, side right |
+----------+----------------------------------------------------+
| 0b0111 | 8 channels: front left, front right, front center, |
| | LFE, back left, back right, side left, side right |
+----------+----------------------------------------------------+
| 0b1000 | 2 channels: left, right; stored as left-side |
| | stereo |
+----------+----------------------------------------------------+
| 0b1001 | 2 channels: left, right; stored as side-right |
| | stereo |
+----------+----------------------------------------------------+
| 0b1010 | 2 channels: left, right; stored as mid-side stereo |
+----------+----------------------------------------------------+
| 0b1011 - | Reserved |
| 0b1111 | |
+----------+----------------------------------------------------+
Table 16
9.1.4. Bit Depth Bits
The next 3 bits (bits 5, 6, and 7 of each fourth byte of each frame)
contain the bit depth of the audio according to the following table.
The next bit is reserved and MUST be zero.
+=======+========================================================+
| Value | Bit Depth |
+=======+========================================================+
| 0b000 | Bit depth only stored in the streaminfo metadata block |
+-------+--------------------------------------------------------+
| 0b001 | 8 bits per sample |
+-------+--------------------------------------------------------+
| 0b010 | 12 bits per sample |
+-------+--------------------------------------------------------+
| 0b011 | Reserved |
+-------+--------------------------------------------------------+
| 0b100 | 16 bits per sample |
+-------+--------------------------------------------------------+
| 0b101 | 20 bits per sample |
+-------+--------------------------------------------------------+
| 0b110 | 24 bits per sample |
+-------+--------------------------------------------------------+
| 0b111 | 32 bits per sample |
+-------+--------------------------------------------------------+
Table 17
9.1.5. Coded Number
Following the reserved bit (starting at the fifth byte of the frame)
is either a sample or a frame number, which will be referred to as
the coded number. When dealing with variable block size streams, the
sample number of the first sample in the frame is encoded. When the
file contains a fixed block size stream, the frame number is encoded.
See Section 9.1 on the blocking strategy bit, which signals whether a
stream is a fixed block size stream or a variable block size stream.
See also Appendix B.1.
The coded number is stored in a variable-length code like UTF-8 as
defined in [RFC3629] but extended to a maximum of 36 bits unencoded
or 7 bytes encoded.
When a frame number is encoded, the value MUST NOT be larger than
what fits a value of 31 bits unencoded or 6 bytes encoded. Please
note that as most general purpose UTF-8 encoders and decoders follow
[RFC3629], they will not be able to handle these extended codes.
Furthermore, while UTF-8 is specifically used to encode characters,
FLAC uses it to encode numbers instead. To encode or decode a coded
number, follow the procedures in Section 3 of [RFC3629], but instead
of using a character number, use a frame or sample number. In
addition, use the extended table below instead of the table in
Section 3 of [RFC3629].
+============================+=====================================+
| Number Range (Hexadecimal) | Octet Sequence (Binary) |
+============================+=====================================+
| 0000 0000 0000 - | 0xxxxxxx |
| 0000 0000 007F | |
+----------------------------+-------------------------------------+
| 0000 0000 0080 - | 110xxxxx 10xxxxxx |
| 0000 0000 07FF | |
+----------------------------+-------------------------------------+
| 0000 0000 0800 - | 1110xxxx 10xxxxxx 10xxxxxx |
| 0000 0000 FFFF | |
+----------------------------+-------------------------------------+
| 0000 0001 0000 - | 11110xxx 10xxxxxx 10xxxxxx 10xxxxxx |
| 0000 001F FFFF | |
+----------------------------+-------------------------------------+
| 0000 0020 0000 - | 111110xx 10xxxxxx 10xxxxxx 10xxxxxx |
| 0000 03FF FFFF | 10xxxxxx |
+----------------------------+-------------------------------------+
| 0000 0400 0000 - | 1111110x 10xxxxxx 10xxxxxx 10xxxxxx |
| 0000 7FFF FFFF | 10xxxxxx 10xxxxxx |
+----------------------------+-------------------------------------+
| 0000 8000 0000 - | 11111110 10xxxxxx 10xxxxxx 10xxxxxx |
| 000F FFFF FFFF | 10xxxxxx 10xxxxxx 10xxxxxx |
+----------------------------+-------------------------------------+
Table 18
If the coded number is a frame number, it MUST be equal to the number
of frames preceding the current frame. If the coded number is a
sample number, it MUST be equal to the number of samples preceding
the current frame. In a stream where these requirements are not met,
seeking is not (reliably) possible.
For example, for a frame that belongs to a variable block size stream
and has exactly 51 billion samples preceding it, the coded number is
constructed as follows:
Octets 1-5
0b11111110 0b10101111 0b10011111 0b10110101 0b10100011
^^^^^^ ^^^^^^ ^^^^^^ ^^^^^^
| | | Bits 18-13
| | Bits 24-19
| Bits 30-25
Bits 36-31
Octets 6-7
0b10111000 0b10000000
^^^^^^ ^^^^^^
| Bits 6-1
Bits 12-7
A decoder that relies on the coded number during seeking could be
vulnerable to buffer overflows or getting stuck in an infinite loop
if it seeks in a stream where the coded numbers are not strictly
increasing or are otherwise not valid. See also Section 11.
9.1.6. Uncommon Block Size
If the block size bits defined earlier in this section are 0b0110 or
0b0111 (uncommon block size minus 1 stored), the block size minus 1
follows the coded number as either an 8-bit or 16-bit unsigned number
coded big-endian. A value of 65535 (corresponding to a block size of
65536) is forbidden and MUST NOT be used, because such a block size
cannot be represented in the streaminfo metadata block. A value from
0 up to (and including) 14, which corresponds to a block size from 1
to 15, is only valid for the last frame in a stream and MUST NOT be
used for any other frame. See also Section 8.2.
9.1.7. Uncommon Sample Rate
If the sample rate bits are 0b1100, 0b1101, or 0b1110 (uncommon
sample rate stored), the sample rate follows the uncommon block size
(or the coded number if no uncommon block size is stored) as either
an 8-bit or a 16-bit unsigned number coded big-endian.
The sample rate MUST NOT be 0 when the subframe contains audio. A
sample rate of 0 MAY be used when non-audio is represented. See
Section 8.2 for details.
9.1.8. Frame Header CRC
Finally, an 8-bit CRC follows the frame/sample number, an uncommon
block size, or an uncommon sample rate (depending on whether the
latter two are stored). This CRC is initialized with 0 and has the
polynomial x^8 + x^2 + x^1 + x^0. This CRC covers the whole frame
header before the CRC, including the sync code.
9.2. Subframes
Following the frame header are a number of subframes equal to the
number of audio channels. Note that subframes contain a bitstream
that does not necessarily have to be a whole number of bytes, so only
the first subframe starts at a byte boundary.
9.2.1. Subframe Header
Each subframe starts with a header. The first bit of the header MUST
be 0, followed by 6 bits that describe which subframe type is used
according to the following table, where v is the value of the 6 bits
as an unsigned number.
+=====================+===========================================+
| Value | Subframe Type |
+=====================+===========================================+
| 0b000000 | Constant subframe |
+---------------------+-------------------------------------------+
| 0b000001 | Verbatim subframe |
+---------------------+-------------------------------------------+
| 0b000010 - 0b000111 | Reserved |
+---------------------+-------------------------------------------+
| 0b001000 - 0b001100 | Subframe with a fixed predictor of order |
| | v-8; i.e., 0, 1, 2, 3 or 4 |
+---------------------+-------------------------------------------+
| 0b001101 - 0b011111 | Reserved |
+---------------------+-------------------------------------------+
| 0b100000 - 0b111111 | Subframe with a linear predictor of order |
| | v-31; i.e., 1 through 32 (inclusive) |
+---------------------+-------------------------------------------+
Table 19
Following the subframe type bits is a bit that flags whether the
subframe uses any wasted bits (see Section 9.2.2). If the flag bit
is 0, the subframe doesn't use any wasted bits and the subframe
header is complete. If the flag bit is 1, the subframe uses wasted
bits and the number of used wasted bits minus 1 appears in unary
form, directly following the flag bit.
9.2.2. Wasted Bits per Sample
Most uncompressed audio file formats can only store audio samples
with a bit depth that is an integer number of bytes. Samples in
which the bit depth is not an integer number of bytes are usually
stored in such formats by padding them with least-significant zero
bits to a bit depth that is an integer number of bytes. For example,
shifting a 14-bit sample right by 2 pads it to a 16-bit sample, which
then has two zero least-significant bits. In this specification,
these least-significant zero bits are referred to as wasted bits per
sample or simply wasted bits. They are wasted in the sense that they
contain no information but are stored anyway.
The FLAC format can optionally take advantage of these wasted bits by
signaling their presence and coding the subframe without them. To do
this, the wasted bits per sample flag in a subframe header is set to
1 and the number of wasted bits per sample (k) minus 1 follows the
flag in an unary encoding. For example, if k is 3, 0b001 follows.
If k = 0, the wasted bits per sample flag is 0 and no unary-coded k
follows. In this document, if a subframe header signals a certain
number of wasted bits, it is said it "uses" these wasted bits.
If a subframe uses wasted bits (i.e., k is not equal to 0), samples
are coded ignoring k least-significant bits. For example, if a frame
not employing stereo decorrelation specifies a sample size of 16 bits
per sample in the frame header and k of a subframe is 3, samples in
the subframe are coded as 13 bits per sample. For more details, see
Section 9.2.3 on how the bit depth of a subframe is calculated. A
decoder MUST add k least-significant zero bits by shifting left
(padding) after decoding a subframe sample. If the frame has left-
side, side-right, or mid-side stereo, a decoder MUST perform padding
on the subframes before restoring the channels to left and right.
The number of wasted bits per sample MUST be such that the resulting
number of bits per sample (of which the calculation is explained in
Section 9.2.3) is larger than zero.
Besides audio files that have a certain number of wasted bits for the
whole file, audio files exist in which the number of wasted bits
varies. There are DVD-Audio discs in which blocks of samples have
had their least-significant bits selectively zeroed to slightly
improve the compression of their otherwise lossless Meridian Lossless
Packing codec; see [MLP]. There are also audio processors like
lossyWAV (see [lossyWAV]) that zero a number of least-significant
bits for a block of samples, increasing the compression in a non-
lossless way. Because of this, the number of wasted bits k MAY
change between frames and MAY differ between subframes. If the
number of wasted bits changes halfway through a subframe (e.g., the
first part has 2 wasted bits and the second part has 4 wasted bits),
the subframe uses the lowest number of wasted bits; otherwise, non-
zero bits would be discarded, and the process would not be lossless.
9.2.3. Constant Subframe
In a constant subframe, only a single sample is stored. This sample
is stored as an integer number coded big-endian, signed two's
complement. The number of bits used to store this sample depends on
the bit depth of the current subframe. The bit depth of a subframe
is equal to the bit depth as coded in the frame header (see
Section 9.1.4) minus the number of used wasted bits coded in the
subframe header (see Section 9.2.2). If a subframe is a side
subframe (see Section 4.2), the bit depth of that subframe is
increased by 1 bit.
9.2.4. Verbatim Subframe
A verbatim subframe stores all samples unencoded in sequential order.
See Section 9.2.3 on how a sample is stored unencoded. The number of
samples that need to be stored in a subframe is provided by the block
size in the frame header.
9.2.5. Fixed Predictor Subframe
Five different fixed predictors are defined in the following table,
one for each prediction order 0 through 4. The table also contains a
derivation that explains the rationale for choosing these fixed
predictors.
+=======+==================================+======================+
| Order | Prediction | Derivation |
+=======+==================================+======================+
| 0 | 0 | N/A |
+-------+----------------------------------+----------------------+
| 1 | a(n-1) | N/A |
+-------+----------------------------------+----------------------+
| 2 | 2 * a(n-1) - a(n-2) | a(n-1) + a'(n-1) |
+-------+----------------------------------+----------------------+
| 3 | 3 * a(n-1) - 3 * a(n-2) + a(n-3) | a(n-1) + a'(n-1) + |
| | | a''(n-1) |
+-------+----------------------------------+----------------------+
| 4 | 4 * a(n-1) - 6 * a(n-2) + 4 * | a(n-1) + a'(n-1) + |
| | a(n-3) - a(n-4) | a''(n-1) + a'''(n-1) |
+-------+----------------------------------+----------------------+
Table 20
Where:
* n is the number of the sample being predicted.
* a(n) is the sample being predicted.
* a(n-1) is the sample before the one being predicted.
* a'(n-1) is the difference between the previous sample and the
sample before that, i.e., a(n-1) - a(n-2). This is the closest
available first-order discrete derivative.
* a''(n-1) is a'(n-1) - a'(n-2) or the closest available second-
order discrete derivative.
* a'''(n-1) is a''(n-1) - a''(n-2) or the closest available third-
order discrete derivative.
As a predictor makes use of samples preceding the sample that is
predicted, it can only be used when enough samples are known. As
each subframe in FLAC is coded completely independently, the first
few samples in each subframe cannot be predicted. Therefore, a
number of so-called warm-up samples equal to the predictor order is
stored. These are stored unencoded, bypassing the predictor and
residual coding stages. See Section 9.2.3 on how samples are stored
unencoded. The table below defines how a fixed predictor subframe
appears in the bitstream.
+==========+===========================================+
| Data | Description |
+==========+===========================================+
| s(n) | Unencoded warm-up samples (n = subframe's |
| | bits per sample * predictor order). |
+----------+-------------------------------------------+
| Coded | Coded residual as defined in |
| residual | Section 9.2.7 |
+----------+-------------------------------------------+
Table 21
Because fixed predictors are specified, they do not have to be
stored. The fixed predictor order, which is stored in the subframe
header, specifies which predictor is used.
To encode a signal with a fixed predictor, each sample has the
corresponding prediction subtracted and sent to the residual coder.
To decode a signal with a fixed predictor, the residual is decoded,
and then the prediction can be added for each sample. This means
that decoding is necessarily a sequential process within a subframe,
as for each sample, enough fully decoded previous samples are needed
to calculate the prediction.
For fixed predictor order 0, the prediction is always 0; thus, each
residual sample is equal to its corresponding input or decoded
sample. The difference between a fixed predictor with order 0 and a
verbatim subframe is that a verbatim subframe stores all samples
unencoded while a fixed predictor with order 0 has all its samples
processed by the residual coder.
The first-order fixed predictor is comparable to how differential
pulse-code modulation (DPCM) encoding works, as the resulting
residual sample is the difference between the corresponding sample
and the sample before it. The higher-order fixed predictors can be
understood as polynomials fitted to the previous samples.
9.2.6. Linear Predictor Subframe
Whereas fixed predictors are well suited for simple signals, using a
(non-fixed) linear predictor on more complex signals can improve
compression by making the residual samples even smaller. There is a
certain trade-off, however, as storing the predictor coefficients
takes up space as well.
In the FLAC format, a predictor is defined by up to 32 predictor
coefficients and a shift. To form a prediction, each coefficient is
multiplied by its corresponding past sample, the results are summed,
and this sum is then shifted. To encode a signal with a linear
predictor, each sample has the corresponding prediction subtracted
and sent to the residual coder. To decode a signal with a linear
predictor, the residual is decoded, and then the prediction can be
added for each sample. This means that decoding MUST be a sequential
process within a subframe, as enough decoded samples are needed to
calculate the prediction for each sample.
The table below defines how a linear predictor subframe appears in
the bitstream.
+==========+==========================================+
| Data | Description |
+==========+==========================================+
| s(n) | Unencoded warm-up samples (n = |
| | subframe's bits per sample * LPC order). |
+----------+------------------------------------------+
| u(4) | (Predictor coefficient precision in |
| | bits)-1 (Note: 0b1111 is forbidden). |
+----------+------------------------------------------+
| s(5) | Prediction right shift needed in bits. |
+----------+------------------------------------------+
| s(n) | Predictor coefficients (n = predictor |
| | coefficient precision * LPC order). |
+----------+------------------------------------------+
| Coded | Coded residual as defined in |
| residual | Section 9.2.7. |
+----------+------------------------------------------+
Table 22
See Section 9.2.3 on how the warm-up samples are stored unencoded.
The predictor coefficients are stored as an integer number coded big-
endian, signed two's complement, where the number of bits needed for
each coefficient is defined by the predictor coefficient precision.
While the prediction right shift is signed two's complement, this
number MUST NOT be negative; see Appendix B.4 for an explanation why
this is.
Please note that the order in which the predictor coefficients appear
in the bitstream corresponds to which *past* sample they belong to.
In other words, the order of the predictor coefficients is opposite
to the chronological order of the samples. So, the first predictor
coefficient has to be multiplied with the sample directly before the
sample that is being predicted, the second predictor coefficient has
to be multiplied with the sample before that, etc.
9.2.7. Coded Residual
The first two bits in a coded residual indicate which coding method
is used. See the table below.
+=============+=============================================+
| Value | Description |
+=============+=============================================+
| 0b00 | Partitioned Rice code with 4-bit parameters |
+-------------+---------------------------------------------+
| 0b01 | Partitioned Rice code with 5-bit parameters |
+-------------+---------------------------------------------+
| 0b10 - 0b11 | Reserved |
+-------------+---------------------------------------------+
Table 23
Both defined coding methods work the same way but differ in the
number of bits used for Rice parameters. The 4 bits that directly
follow the coding method bits form the partition order, which is an
unsigned number. The rest of the coded residual consists of
2^(partition order) partitions. For example, if the 4 bits are
0b1000, the partition order is 8, and the residual is split up into
2^8 = 256 partitions.
Each partition contains a certain number of residual samples. The
number of residual samples in the first partition is equal to (block
size >> partition order) - predictor order, i.e., the block size
divided by the number of partitions minus the predictor order. In
all other partitions, the number of residual samples is equal to
(block size >> partition order).
The partition order MUST be such that the block size is evenly
divisible by the number of partitions. This means, for example, that
only partition order 0 is allowed for all odd block sizes. The
partition order also MUST be such that the (block size >> partition
order) is larger than the predictor order. This means, for example,
that with a block size of 4096 and a predictor order of 4, the
partition order cannot be larger than 9.
Each partition starts with a parameter. If the coded residual of a
subframe is one with 4-bit Rice parameters (see Table 23), the first
4 bits of each partition are either a Rice parameter or an escape
code. These 4 bits indicate an escape code if they are 0b1111;
otherwise, they contain the Rice parameter as an unsigned number. If
the coded residual of the current subframe is one with 5-bit Rice
parameters, the first 5 bits of each partition indicate an escape
code if they are 0b11111; otherwise, they contain the Rice parameter
as an unsigned number as well.
9.2.7.1. Escaped Partition
If an escape code was used, the partition does not contain a
variable-length Rice-coded residual; rather, it contains a fixed-
length unencoded residual. Directly following the escape code are 5
bits containing the number of bits with which each residual sample is
stored, as an unsigned number. The residual samples themselves are
stored signed two's complement. For example, when a partition is
escaped and each residual sample is stored with 3 bits, the number -1
is represented as 0b111.
Note that it is possible that the number of bits with which each
sample is stored is 0, which means that all residual samples in that
partition have a value of 0 and that no bits are used to store the
samples. In that case, the partition contains nothing except the
escape code and 0b00000.
9.2.7.2. Rice Code
If a Rice parameter was provided for a certain partition, that
partition contains a Rice-coded residual. The residual samples,
which are signed numbers, are represented by unsigned numbers in the
Rice code. For positive numbers, the representation is the number
doubled. For negative numbers, the representation is the number
multiplied by -2 and with 1 subtracted. This representation of
signed numbers is also known as zigzag encoding. The zigzag-encoded
residual is called the folded residual.
Each folded residual sample is then split into two parts, a most-
significant part and a least-significant part. The Rice parameter at
the start of each partition determines where that split lies: it is
the number of bits in the least-significant part. Each residual
sample is then stored by coding the most-significant part as unary,
followed by the least-significant part as binary.
For example, take a partition with Rice parameter 3 containing a
folded residual sample with 38 as its value, which is 0b100110 in
binary. The most-significant part is 0b100 (4) and is stored in
unary form as 0b00001. The least-significant part is 0b110 (6) and
is stored as is. The Rice code word is thus 0b00001110. The Rice
code words for all residual samples in a partition are stored
consecutively.
To decode a Rice code word, zero bits must be counted until
encountering a one bit, after which a number of bits given by the
Rice parameter must be read. The count of zero bits is shifted left
by the Rice parameter (i.e., multiplied by 2 raised to the power Rice
parameter) and bitwise ORed with (i.e., added to) the read value.
This is the folded residual value. An even folded residual value is
shifted right 1 bit (i.e., divided by 2) to get the (unfolded)
residual value. An odd folded residual value is shifted right 1 bit
and then has all bits flipped (1 added to and divided by -2) to get
the (unfolded) residual value, subject to negative numbers being
signed two's complement on the decoding machine.
Appendix D shows decoding of a complete coded residual.
9.2.7.3. Residual Sample Value Limit
All residual sample values MUST be representable in the range offered
by a 32-bit integer, signed one's complement. Equivalently, all
residual sample values MUST fall in the range offered by a 32-bit
integer signed two's complement, excluding the most negative possible
value of that range. This means residual sample values MUST NOT have
an absolute value equal to, or larger than, 2 to the power 31. A
FLAC encoder MUST make sure of this. If a FLAC encoder is, for a
certain subframe, unable to find a suitable predictor for which all
residual samples fall within said range, it MUST default to writing a
verbatim subframe. Appendix A explains in which circumstances
residual samples are already implicitly representable in said range;
thus, an additional check is not needed.
The reason for this limit is to ensure that decoders can use 32-bit
integers when processing residuals, simplifying decoding. The reason
the most negative value of a 32-bit integer signed two's complement
is specifically excluded is to prevent decoders from having to
implement specific handling of that value, as it cannot be negated
within a 32-bit signed integer, and most library routines calculating
an absolute value have undefined behavior for processing that value.
9.3. Frame Footer
Following the last subframe is the frame footer. If the last
subframe is not byte aligned (i.e., the number of bits required to
store all subframes put together is not divisible by 8), zero bits
are added until byte alignment is reached. Following this is a
16-bit CRC, initialized with 0, with the polynomial x^16 + x^15 + x^2
+ x^0. This CRC covers the whole frame, excluding the 16-bit CRC but
including the sync code.
10. Container Mappings
The FLAC format can be used without any container, as it already
provides for the most basic features normally associated with a
container. However, the functionality this basic container provides
is rather limited, and for more advanced features (such as combining
FLAC audio with video), it needs to be encapsulated by a more capable
container. This presents a problem: because of these container
features, the FLAC format mixes data that belongs to the encoded data
(like block size and sample rate) with data that belongs to the
container (like checksum and timecode). The choice was made to
encapsulate FLAC frames as they are, which means some data will be
duplicated and potentially deviating between the FLAC frames and the
encapsulating container.
As FLAC frames are completely independent of each other, container
format features handling dependencies do not need to be used. For
example, all FLAC frames embedded in Matroska are marked as keyframes
when they are stored in a SimpleBlock, and tracks in an MP4 file
containing only FLAC frames do not need a sync sample box.
10.1. Ogg Mapping
The Ogg container format is defined in [RFC3533]. The first packet
of a logical bitstream carrying FLAC data is structured according to
the following table.
+=========+=========================================================+
| Data | Description |
+=========+=========================================================+
| 5 | Bytes 0x7F 0x46 0x4C 0x41 0x43 (as also defined by |
| bytes | [RFC5334]). |
+---------+---------------------------------------------------------+
| 2 | Version number of the FLAC-in-Ogg mapping. These bytes |
| bytes | are 0x01 0x00, meaning version 1.0 of the mapping. |
+---------+---------------------------------------------------------+
| 2 | Number of header packets (excluding the first header |
| bytes | packet) as an unsigned number coded big-endian. |
+---------+---------------------------------------------------------+
| 4 | The fLaC signature. |
| bytes | |
+---------+---------------------------------------------------------+
| 4 | A metadata block header for the streaminfo metadata |
| bytes | block. |
+---------+---------------------------------------------------------+
| 34 | A streaminfo metadata block. |
| bytes | |
+---------+---------------------------------------------------------+
Table 24
The number of header packets MAY be 0, which means the number of
packets that follow is unknown. This first packet MUST NOT share a
Ogg page with any other packets. This means the first page of a
logical stream of FLAC-in-Ogg is always 79 bytes.
Following the first packet are one or more header packets, each of
which contains a single metadata block. The first of these packets
SHOULD be a Vorbis comment metadata block for historic reasons. This
is contrary to unencapsulated FLAC streams, where the order of
metadata blocks is not important except for the streaminfo metadata
block and where a Vorbis comment metadata block is optional.
Following the header packets are audio packets. Each audio packet
contains a single FLAC frame. The first audio packet MUST start on a
new Ogg page, i.e., the last metadata block MUST finish its page
before any audio packets are encapsulated.
The granule position of all pages containing header packets MUST be
0. For pages containing audio packets, the granule position is the
number of the last sample contained in the last completed packet in
the frame. The sample numbering considers interchannel samples. If
a page contains no packet end (e.g., when it only contains the start
of a large packet that continues on the next page), then the granule
position is set to the maximum value possible, i.e., 0xFF 0xFF 0xFF
0xFF 0xFF 0xFF 0xFF 0xFF.
The granule position of the first audio data page with a completed
packet MAY be larger than the number of samples contained in packets
that complete on that page. In other words, the apparent sample
number of the first sample in the stream following from the granule
position and the audio data MAY be larger than 0. This allows, for
example, a server to cast a live stream to several clients that
joined at different moments without rewriting the granule position
for each client.
If an audio stream is encoded where audio properties (sample rate,
number of channels, or bit depth) change at some point in the stream,
this should be dealt with by finishing encoding of the current Ogg
stream and starting a new Ogg stream, concatenated to the previous
one. This is called chaining in Ogg. See the Ogg specification
[RFC3533] for details.
10.2. Matroska Mapping
The Matroska container format is defined in [RFC9559]. The codec ID
(EBML path \Segment\Tracks\TrackEntry\CodecID) assigned to signal
tracks carrying FLAC data is A_FLAC in ASCII. All FLAC data before
the first audio frame (i.e., the fLaC ASCII signature and all
metadata blocks) is stored as CodecPrivate data (EBML path
\Segment\Tracks\TrackEntry\CodecPrivate).
Each FLAC frame (including all of its subframes) is treated as a
single frame in the context of Matroska.
If an audio stream is encoded where audio properties (sample rate,
number of channels, or bit depth) change at some point in the stream,
this should be dealt with by finishing the current Matroska segment
and starting a new one with the new properties.
10.3. ISO Base Media File Format (MP4) Mapping
The full encapsulation definition of FLAC audio in MP4 files was
deemed too extensive to include in this document. A definition
document can be found at [FLAC-in-MP4-specification].
11. Security Considerations
Like any other codec (such as [RFC6716]), FLAC should not be used
with insecure ciphers or cipher modes that are vulnerable to known
plaintext attacks. Some of the header bits, as well as the padding,
are easily predictable.
Implementations of the FLAC codec need to take appropriate security
considerations into account. Section 2.1 of [RFC4732] provides
general information on DoS attacks on end systems and describes some
mitigation strategies. Areas of concern specific to FLAC follow.
It is extremely important for the decoder to be robust against
malformed payloads. Payloads that do not conform to this
specification MUST NOT cause the decoder to overrun its allocated
memory or take an excessive amount of resources to decode. An
overrun in allocated memory could lead to arbitrary code execution by
an attacker. The same applies to the encoder, even though problems
with encoders are typically rarer. Malformed audio streams MUST NOT
cause the encoder to misbehave because this would allow an attacker
to attack transcoding gateways.
As with all compression algorithms, both encoding and decoding can
produce an output much larger than the input. For decoding, the most
extreme possible case of this is a frame with eight constant
subframes of block size 65535 and coding for 32-bit PCM. This frame
is only 49 bytes in size but codes for more than 2 megabytes of
uncompressed PCM data. For encoding, it is possible to have an even
larger size increase, although such behavior is generally considered
faulty. This happens if the encoder chooses a Rice parameter that
does not fit with the residual that has to be encoded. In such a
case, very long unary-coded symbols can appear (in the most extreme
case, more than 4 gigabytes per sample). Decoder and encoder
implementors are advised to take precautions to prevent excessive
resource utilization in such cases.
Where metadata is handled, implementors are advised to either
thoroughly test the handling of extreme cases or impose reasonable
limits beyond the limits of this specification. For example, a
single Vorbis comment metadata block can contain millions of valid
fields. It is unlikely such a limit is ever reached except in a
potentially malicious file. Likewise, the media type and description
of a picture metadata block can be millions of characters long,
despite there being no reasonable use of such contents. One possible
use case for very long character strings is in lyrics, which can be
stored in Vorbis comment metadata block fields.
Various kinds of metadata blocks contain length fields or field
counts. While reading a block following these lengths or counts, a
decoder MUST make sure higher-level lengths or counts (most
importantly, the length field of the metadata block itself) are not
exceeded. As some of these length fields code string lengths and
memory must be allocated for that, parsers MUST first verify that a
block is valid before allocating memory based on its contents, except
when explicitly instructed to salvage data from a malformed file.
Metadata blocks can also contain references, e.g., the picture
metadata block can contain a URI. When following a URI, the security
considerations of [RFC3986] apply. Applications MUST obtain explicit
user approval to retrieve resources via remote protocols. Following
external URIs introduces a tracking risk from on-path observers and
the operator of the service hosting the URI. Likewise, the choice of
scheme, if it isn't protected like https, could also introduce
integrity attacks by an on-path observer. A malicious operator of
the service hosting the URI can return arbitrary content that the
parser will read. Also, such retrievals can be used in a DDoS attack
when the URI points to a potential victim. Therefore, applications
need to ask user approval for each retrieval individually, take extra
precautions when parsing retrieved data, and cache retrieved
resources. Applications MUST obtain explicit user approval to
retrieve local resources not located in the same directory as the
FLAC file being processed. Since relative URIs are permitted,
applications MUST guard against directory traversal attacks and guard
against a violation of a same-origin policy if such a policy is being
enforced.
Seeking in a FLAC stream that is not in a container relies on the
coded number in frame headers and optionally a seek table metadata
block. Parsers MUST employ thorough checks on whether a found coded
number or seek point is at all possible, e.g., whether it is within
bounds and not directly contradicting any other coded number or seek
point that the seeking process relies on. Without these checks,
seeking might get stuck in an infinite loop when numbers in frames
are non-consecutive or otherwise not valid, which could be used in
DoS attacks.
Implementors are advised to employ fuzz testing combined with
different sanitizers on FLAC decoders to find security problems.
Ignoring the results of CRC checks improves the efficiency of decoder
fuzz testing.
See [FLAC-decoder-testbench] for a non-exhaustive list of FLAC files
with extreme configurations that lead to crashes or reboots on some
known implementations. Besides providing a starting point for
security testing, this set of files can also be used to test
conformance with this specification.
FLAC files may contain executable code, although the FLAC format is
not designed for it and it is uncommon. One use case where FLAC is
occasionally used to store executable code is when compressing images
of mixed-mode CDs, which contain both audio and non-audio data, the
non-audio portion of which can contain executable code. In that
case, the executable code is stored as if it were audio and is
potentially obscured. Of course, it is also possible to store
executable code as metadata, for example, as a Vorbis comment with
help of a binary-to-text encoding or directly in an application
metadata block. Applications MUST NOT execute code contained in FLAC
files or present parts of FLAC files as executable code to the user,
except when an application has that explicit purpose, e.g.,
applications reading FLAC files as disc images and presenting it as a
virtual disc drive.
12. IANA Considerations
Per this document, IANA has registered one new media type ("audio/
flac") and created a new IANA registry, as described in the
subsections below.
12.1. Media Type Registration
IANA has registered the "audio/flac" media type as follows. This
media type is applicable for FLAC audio that is not packaged in a
container as described in Section 10. FLAC audio packaged in such a
container will take on the media type of that container, for example,
"audio/ogg" when packaged in an Ogg container or "video/mp4" when
packaged in an MP4 container alongside a video track.
Type name: audio
Subtype name: flac
Required parameters: N/A
Optional parameters: N/A
Encoding considerations: as per RFC 9639
Security considerations: See the security considerations in
Section 11 of RFC 9639.
Interoperability considerations: See the descriptions of past format
changes in Appendix B of RFC 9639.
Published specification: RFC 9639
Applications that use this media type: FFmpeg, Apache, Firefox
Fragment identifier considerations: N/A
Additional information:
Deprecated alias names for this type: audio/x-flac
Magic number(s): fLaC
File extension(s): flac
Macintosh file type code(s): N/A
Uniform Type Identifier: org.xiph.flac conforms to public.audio
Windows Clipboard Format Name: audio/flac
Person & email address to contact for further information: IETF
CELLAR Working Group (cellar@ietf.org)
Intended usage: COMMON
Restrictions on usage: N/A
Author: IETF CELLAR Working Group
Change controller: Internet Engineering Task Force (iesg@ietf.org)
12.2. FLAC Application Metadata Block IDs Registry
IANA has created a new registry called the "FLAC Application Metadata
Block IDs" registry. The values correspond to the 32-bit identifier
described in Section 8.4.
To register a new application ID in this registry, one needs an
application ID, a description, an optional reference to a document
describing the application ID, and a Change Controller (IETF or email
of registrant). The application IDs are allocated according to the
"First Come First Served" policy [RFC8126] so that there is no
impediment to registering any application IDs the FLAC community
encounters, especially if they were used in audio files but were not
registered when the audio files were encoded. An application ID can
be any 32-bit value but is often composed of 4 ASCII characters that
are human-readable.
The initial contents of "FLAC Application Metadata Block IDs"
registry are shown in the table below. These initial values were
taken from the registration page at xiph.org (see
[ID-registration-page]), which is no longer being maintained as it
has been replaced by this registry.
+===========+==========+===========+===================+==========+
|Application|ASCII |Description|Reference |Change |
|ID |Rendition | | |Controller|
| |(If | | | |
| |Available)| | | |
+===========+==========+===========+===================+==========+
|0x41544348 |ATCH |FlacFile |[FlacFile], RFC |IETF |
| | | |9639 | |
+-----------+----------+-----------+-------------------+----------+
|0x42534F4C |BSOL |beSolo |RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x42554753 |BUGS |Bugs Player|RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x43756573 |Cues |GoldWave |RFC 9639 |IETF |
| | |cue points | | |
+-----------+----------+-----------+-------------------+----------+
|0x46696361 |Fica |CUE |RFC 9639 |IETF |
| | |Splitter | | |
+-----------+----------+-----------+-------------------+----------+
|0x46746F6C |Ftol |flac-tools |RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x4D4F5442 |MOTB |MOTB |RFC 9639 |IETF |
| | |MetaCzar | | |
+-----------+----------+-----------+-------------------+----------+
|0x4D505345 |MPSE |MP3 Stream |RFC 9639 |IETF |
| | |Editor | | |
+-----------+----------+-----------+-------------------+----------+
|0x4D754D4C |MuML |MusicML: |RFC 9639 |IETF |
| | |Music | | |
| | |Metadata | | |
| | |Language | | |
+-----------+----------+-----------+-------------------+----------+
|0x52494646 |RIFF |Sound |RFC 9639 |IETF |
| | |Devices | | |
| | |RIFF chunk | | |
| | |storage | | |
+-----------+----------+-----------+-------------------+----------+
|0x5346464C |SFFL |Sound Font |RFC 9639 |IETF |
| | |FLAC | | |
+-----------+----------+-----------+-------------------+----------+
|0x534F4E59 |SONY |Sony |RFC 9639 |IETF |
| | |Creative | | |
| | |Software | | |
+-----------+----------+-----------+-------------------+----------+
|0x5351455A |SQEZ |flacsqueeze|RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x54745776 |TtWv |TwistedWave|RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x55495453 |UITS |UITS |RFC 9639 |IETF |
| | |Embedding | | |
| | |tools | | |
+-----------+----------+-----------+-------------------+----------+
|0x61696666 |aiff |FLAC AIFF |[Foreign-metadata],|IETF |
| | |chunk |RFC 9639 | |
| | |storage | | |
+-----------+----------+-----------+-------------------+----------+
|0x696D6167 |imag |flac-image |RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x7065656D |peem |Parseable |RFC 9639 |IETF |
| | |Embedded | | |
| | |Extensible | | |
| | |Metadata | | |
+-----------+----------+-----------+-------------------+----------+
|0x71667374 |qfst |QFLAC |RFC 9639 |IETF |
| | |Studio | | |
+-----------+----------+-----------+-------------------+----------+
|0x72696666 |riff |FLAC RIFF |[Foreign-metadata],|IETF |
| | |chunk |RFC 9639 | |
| | |storage | | |
+-----------+----------+-----------+-------------------+----------+
|0x74756E65 |tune |TagTuner |RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x77363420 |w64 |FLAC Wave64|[Foreign-metadata],|IETF |
| | |chunk |RFC 9639 | |
| | |storage | | |
+-----------+----------+-----------+-------------------+----------+
|0x78626174 |xbat |XBAT |RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
|0x786D6364 |xmcd |xmcd |RFC 9639 |IETF |
+-----------+----------+-----------+-------------------+----------+
Table 25
13. References
13.1. Normative References
[ISRC-handbook]
International ISRC Registration Authority, "International
Standard Recording Code (ISRC) Handbook", 4th edition,
2021, <https://www.ifpi.org/isrc_handbook/>.
[RFC1321] Rivest, R., "The MD5 Message-Digest Algorithm", RFC 1321,
DOI 10.17487/RFC1321, April 1992,
<https://www.rfc-editor.org/info/rfc1321>.
[RFC2046] Freed, N. and N. Borenstein, "Multipurpose Internet Mail
Extensions (MIME) Part Two: Media Types", RFC 2046,
DOI 10.17487/RFC2046, November 1996,
<https://www.rfc-editor.org/info/rfc2046>.
[RFC2083] Boutell, T., "PNG (Portable Network Graphics)
Specification Version 1.0", RFC 2083,
DOI 10.17487/RFC2083, March 1997,
<https://www.rfc-editor.org/info/rfc2083>.
[RFC2119] Bradner, S., "Key words for use in RFCs to Indicate
Requirement Levels", BCP 14, RFC 2119,
DOI 10.17487/RFC2119, March 1997,
<https://www.rfc-editor.org/info/rfc2119>.
[RFC3533] Pfeiffer, S., "The Ogg Encapsulation Format Version 0",
RFC 3533, DOI 10.17487/RFC3533, May 2003,
<https://www.rfc-editor.org/info/rfc3533>.
[RFC3629] Yergeau, F., "UTF-8, a transformation format of ISO
10646", STD 63, RFC 3629, DOI 10.17487/RFC3629, November
2003, <https://www.rfc-editor.org/info/rfc3629>.
[RFC3986] Berners-Lee, T., Fielding, R., and L. Masinter, "Uniform
Resource Identifier (URI): Generic Syntax", STD 66,
RFC 3986, DOI 10.17487/RFC3986, January 2005,
<https://www.rfc-editor.org/info/rfc3986>.
[RFC8174] Leiba, B., "Ambiguity of Uppercase vs Lowercase in RFC
2119 Key Words", BCP 14, RFC 8174, DOI 10.17487/RFC8174,
May 2017, <https://www.rfc-editor.org/info/rfc8174>.
[RFC9559] Lhomme, S., Bunkus, M., and D. Rice, "Matroska Media
Container Format Specification", RFC 9559,
DOI 10.17487/RFC9559, October 2024,
<https://www.rfc-editor.org/info/rfc9559>.
13.2. Informative References
[Durbin] Durbin, J., "The Fitting of Time-Series Models", Revue de
l'Institut International de Statistique / Review of the
International Statistical Institute, vol. 28, no. 3, pp.
233-44, DOI 10.2307/1401322, 1960,
<https://www.jstor.org/stable/1401322>.
[FIR] Wikipedia, "Finite impulse response", August 2024,
<https://en.wikipedia.org/w/
index.php?title=Finite_impulse_response&oldid=1240945295>.
[FLAC-decoder-testbench]
"The Free Lossless Audio Codec (FLAC) test files", commit
aa7b0c6, August 2023,
<https://github.com/ietf-wg-cellar/flac-test-files>.
[FLAC-implementation]
"FLAC", <https://xiph.org/flac/>.
[FLAC-in-MP4-specification]
"Encapsulation of FLAC in ISO Base Media File Format",
commit 78d85dd, July 2022,
<https://github.com/xiph/flac/blob/master/doc/
isoflac.txt>.
[FLAC-specification-github]
"The Free Lossless Audio Codec (FLAC) Specification",
<https://github.com/ietf-wg-cellar/flac-specification>.
[FLAC-wiki-interoperability]
"Interoperability considerations", commit 58a06d6,
<https://github.com/ietf-wg-cellar/flac-
specification/wiki/Interoperability-considerations>.
[FlacFile] "FlacFile", Wayback Machine archive, October 2007,
<https://web.archive.org/web/20071023070305/
http://firestuff.org:80/flacfile/>.
[Foreign-metadata]
"Specification of foreign metadata storage in FLAC",
commit 72787c3, November 2023,
<https://github.com/xiph/flac/blob/master/doc/
foreign_metadata_storage.md>.
[ID-registration-page]
Xiph.Org, "ID registry", <https://xiph.org/flac/id.html>.
[ID3v2] Nilsson, M., "ID3 tag version 2.4.0 - Native Frames",
Wayback Machine archive, November 2000,
<https://web.archive.org/web/20220903174949/
https://id3.org/id3v2.4.0-frames>.
[IEC.60908.1999]
International Electrotechnical Commission, "Audio
recording - Compact disc digital audio system",
IEC 60908:1999-02, 1999,
<https://webstore.iec.ch/publication/3885>.
[LinearPrediction]
Wikipedia, "Linear prediction", August 2023,
<https://en.wikipedia.org/w/
index.php?title=Linear_prediction&oldid=1169015573>.
[Lossless-Compression]
Hans, M. and R. W. Schafer, "Lossless compression of
digital audio", IEEE Signal Processing Magazine, vol. 18,
no. 4, pp. 21-32, DOI 10.1109/79.939834, July 2001,
<https://ieeexplore.ieee.org/document/939834>.
[lossyWAV] Hydrogenaudio Knowledgebase, "lossyWAV", July 2021,
<https://wiki.hydrogenaud.io/
index.php?title=LossyWAV&oldid=32877>.
[MLP] Gerzon, M. A., Craven, P. G., Stuart, J. R., Law, M. J.,
and R. J. Wilson, "The MLP Lossless Compression System",
Audio Engineering Society Conference: 17th International
Conference: High-Quality Audio Codin, September 1999,
<https://www.aes.org/e-lib/online/browse.cfm?elib=8082>.
[MusicBrainz]
MusicBrainz, "Tags & Variables", MusicBrainz Picard v2.10
documentation, <https://picard-
docs.musicbrainz.org/en/variables/variables.html>.
[RFC4732] Handley, M., Ed., Rescorla, E., Ed., and IAB, "Internet
Denial-of-Service Considerations", RFC 4732,
DOI 10.17487/RFC4732, December 2006,
<https://www.rfc-editor.org/info/rfc4732>.
[RFC5334] Goncalves, I., Pfeiffer, S., and C. Montgomery, "Ogg Media
Types", RFC 5334, DOI 10.17487/RFC5334, September 2008,
<https://www.rfc-editor.org/info/rfc5334>.
[RFC6716] Valin, JM., Vos, K., and T. Terriberry, "Definition of the
Opus Audio Codec", RFC 6716, DOI 10.17487/RFC6716,
September 2012, <https://www.rfc-editor.org/info/rfc6716>.
[RFC8126] Cotton, M., Leiba, B., and T. Narten, "Guidelines for
Writing an IANA Considerations Section in RFCs", BCP 26,
RFC 8126, DOI 10.17487/RFC8126, June 2017,
<https://www.rfc-editor.org/info/rfc8126>.
[Rice] Rice, R. F. and J. R. Plaunt, "Adaptive Variable-Length
Coding for Efficient Compression of Spacecraft Television
Data", IEEE Transactions on Communication Technology, vol.
19, no. 6, pp. 889-897, DOI 10.1109/TCOM.1971.1090789,
December 1971,
<https://ieeexplore.ieee.org/document/1090789>.
[Robinson-TR156]
Robinson, T., "SHORTEN: Simple lossless and near-lossless
waveform compression", Cambridge University Engineering
Department Technical Report CUED/F-INFENG/TR.156, December
1994, <https://mi.eng.cam.ac.uk/reports/svr-ftp/auto-pdf/
robinson_tr156.pdf>.
[Shannon] Shannon, C. E., "Communication in the Presence of Noise",
Proceedings of the IRE, vol. 37, no. 1, pp. 10-21,
DOI 10.1109/JRPROC.1949.232969, January 1949,
<https://ieeexplore.ieee.org/document/1697831>.
[VarLengthCode]
Wikipedia, "Variable-length code", April 2024,
<https://en.wikipedia.org/w/index.php?title=Variable-
length_code&oldid=1220260423>.
[Vorbis] Xiph.Org, "Ogg Vorbis I format specification: comment
field and header specification",
<https://xiph.org/vorbis/doc/v-comment.html>.
Appendix A. Numerical Considerations
In order to maintain lossless behavior, all arithmetic used in
encoding and decoding sample values must be done with integer data
types to eliminate the possibility of introducing rounding errors
associated with floating-point arithmetic. Use of floating-point
representations in analysis (e.g., finding a good predictor or Rice
parameter) is not a concern as long as the process of using the found
predictor and Rice parameter to encode audio samples is implemented
with only integer math.
Furthermore, the possibility of integer overflow can be eliminated by
using data types that are large enough. Choosing a 64-bit signed
data type for all arithmetic involving sample values would make sure
the possibility for overflow is eliminated, but usually, smaller data
types are chosen for increased performance, especially in embedded
devices. This appendix provides guidelines for choosing the
appropriate data type for each step of encoding and decoding FLAC
files.
In this appendix, signed data types are signed two's complement.
A.1. Determining the Necessary Data Type Size
To find the smallest data type size that is guaranteed not to
overflow for a certain sequence of arithmetic operations, the
combination of values producing the largest possible result should be
considered.
For example, if two 16-bit signed integers are added, the largest
possible result forms if both values are the largest number that can
be represented with a 16-bit signed integer. To store the result, a
signed integer data type with at least 17 bits is needed. Similarly,
when adding 4 of these values, 18 bits are needed; when adding 8, 19
bits are needed, etc. In general, the number of bits necessary when
adding numbers together is increased by the log base 2 of the number
of values rounded up to the nearest integer. So, when adding 18
unknown values stored in 8-bit signed integers, we need a signed
integer data type of at least 13 bits to store the result, as the log
base 2 of 18 rounded up is 5.
When multiplying two numbers, the number of bits needed for the
result is the size of the first number plus the size of the second
number. For example, if a 16-bit signed integer is multiplied by
another 16-bit signed integer, the result needs at least 32 bits to
be stored without overflowing. To show this in practice, the largest
signed value that can be stored in 4 bits is -8. (-8)*(-8) is 64,
which needs at least 8 bits (signed) to store.
A.2. Stereo Decorrelation
When stereo decorrelation is used, the side channel will have one
extra bit of bit depth; see Section 4.2.
This means that while 16-bit signed integers have sufficient range to
store samples from a fully decoded FLAC frame with a bit depth of 16
bits, the decoding of a side subframe in such a file will need a data
type with at least 17 bits to store decoded subframe samples before
undoing stereo decorrelation.
Most FLAC decoders store decoded (subframe) samples as 32-bit values,
which is sufficient for files with bit depths up to (and including)
31 bits.
A.3. Prediction
A prediction (which is used to calculate the residual on encoding or
added to the residual to calculate the sample value on decoding) is
formed by multiplying and summing preceding sample values. In order
to eliminate the possibility of integer overflow, the combination of
preceding sample values and predictor coefficients producing the
largest possible value should be considered.
To determine the size of the data type needed to calculate either a
residual sample (on encoding) or an audio sample value (on decoding)
in a fixed predictor subframe, the maximum possible value for these
is calculated as described in Appendix A.1 and in the following
table. For example, if a frame codes for 16-bit audio and has some
form of stereo decorrelation, the subframe coding for the side
channel would need 16+1+3 bits if a third-order fixed predictor is
used.
+=======+==============================+===============+=======+
| Order | Calculation of Residual | Sample Values | Extra |
| | | Summed | Bits |
+=======+==============================+===============+=======+
| 0 | a(n) | 1 | 0 |
+-------+------------------------------+---------------+-------+
| 1 | a(n) - a(n-1) | 2 | 1 |
+-------+------------------------------+---------------+-------+
| 2 | a(n) - 2 * a(n-1) + a(n-2) | 4 | 2 |
+-------+------------------------------+---------------+-------+
| 3 | a(n) - 3 * a(n-1) + 3 * | 8 | 3 |
| | a(n-2) - a(n-3) | | |
+-------+------------------------------+---------------+-------+
| 4 | a(n) - 4 * a(n-1) + 6 * | 16 | 4 |
| | a(n-2) - 4 * a(n-3) + a(n-4) | | |
+-------+------------------------------+---------------+-------+
Table 26
Where:
* n is the number of the sample being predicted.
* a(n) is the sample being predicted.
* a(n-1) is the sample before the one being predicted, a(n-2) is the
sample before that, etc.
For subframes with a linear predictor, the calculation is a little
more complicated. Each prediction is the sum of several
multiplications. Each of these multiply a sample value with a
predictor coefficient. The extra bits needed can be calculated by
adding the predictor coefficient precision (in bits) to the bit depth
of the audio samples. To account for the summing of these
multiplications, the log base 2 of the predictor order rounded up is
added.
For example, if the sample bit depth of the source is 24, the current
subframe encodes a side channel (see Section 4.2), the predictor
order is 12, and the predictor coefficient precision is 15 bits, the
minimum required size of the used signed integer data type is at
least (24 + 1) + 15 + ceil(log2(12)) = 44 bits. As another example,
with a side-channel subframe bit depth of 16, a predictor order of 8,
and a predictor coefficient precision of 12 bits, the minimum
required size of the used signed integer data type is (16 + 1) + 12 +
ceil(log2(8)) = 32 bits.
A.4. Residual
As stated in Section 9.2.7, an encoder must make sure residual
samples are representable by a 32-bit integer, signed two's
complement, excluding the most negative value. As in the previous
section, it is possible to calculate when residual samples already
implicitly fit and when an additional check is needed. This implicit
fit is achieved when residuals would fit a theoretical 31-bit signed
integer, as that satisfies both of the mentioned criteria. When this
implicit fit is not achieved, all residual values must be calculated
and checked individually.
For the residual of a fixed predictor, the maximum residual sample
size was already calculated in the previous section. However, for a
linear predictor, the prediction is shifted right by a certain
amount. The number of bits needed for the residual is the number of
bits calculated in the previous section, reduced by the prediction
right shift, and increased by one bit to account for the subtraction
of the prediction from the current sample on encoding.
Taking the last example of the previous section, where 32 bits were
needed for the prediction, the required data type size for the
residual samples in case of a right shift of 10 bits would be 32 - 10
+ 1 = 23 bits, which means it is not necessary to perform the
aforementioned check.
As another example, when encoding 32-bit PCM with fixed predictors,
all predictor orders must be checked. While the zero-order fixed
predictor is guaranteed to have residual samples that fit a 32-bit
signed integer, it might produce a residual sample value that is the
most negative representable value of that 32-bit signed integer.
Note that on decoding, while the residual sample values are limited
to the aforementioned range, the predictions are not. This means
that while the decoding of the residual samples can happen fully in
32-bit signed integers, decoders must be sure to execute the addition
of each residual sample to its accompanying prediction with a signed
integer data type that is wide enough, as with encoding.
A.5. Rice Coding
When folding (i.e., zigzag encoding) the residual sample values, no
extra bits are needed when the absolute value of each residual sample
is first stored in an unsigned data type of the size of the last
step, then doubled, and then has one subtracted depending on whether
the residual sample was positive or negative. However, many
implementations choose to require one extra bit of data type size so
zigzag encoding can happen in one step without a cast instead of the
procedure described in the previous sentence.
Appendix B. Past Format Changes
This informational appendix documents the changes made to the FLAC
format over the years. This information might be of use when
encountering FLAC files that were made with software following the
format as it was before the changes documented in this appendix.
The FLAC format was first specified in December 2000, and the
bitstream format was considered frozen with the release of FLAC 1.0
(the reference encoder/decoder) in July 2001. Only changes made
since this first stable release are considered in this appendix.
Changes made to the FLAC streamable subset definition (see Section 7)
are not considered.
B.1. Addition of Blocking Strategy Bit
Perhaps the largest backwards-incompatible change to the
specification was published in July 2007. Before this change,
variable block size streams were not explicitly marked as such by a
flag bit in the frame header. A decoder had two ways to detect a
variable block size stream: by comparing the minimum and maximum
block sizes in the streaminfo metadata block (which are equal for a
fixed block size stream) or by detecting a change of block size
during a stream if a decoder did not receive a streaminfo metadata
block, which could not happen at all in theory. As the meaning of
the coded number in the frame header depends on whether or not a
stream has a variable block size, this presented a problem: the
meaning of the coded number could not be reliably determined. To fix
this problem, one of the reserved bits was changed to be used as a
blocking strategy bit. See also Section 9.1.
Along with the addition of a new flag, the meaning of the block size
bits (see Section 9.1.1) was subtly changed. Initially, block size
bits patterns 0b0001-0b0101 and 0b1000-0b1111 could only be used for
fixed block size streams, while 0b0110 and 0b0111 could be used for
both fixed block size and variable block size streams. With this
change, these restrictions were lifted, and patterns 0b0001-0b1111
are now used for both variable block size and fixed block size
streams.
B.2. Restriction of Encoded Residual Samples
Another change to the specification was deemed necessary during
standardization by the CELLAR Working Group of the IETF. As
specified in Section 9.2.7, a limit is imposed on residual samples.
This limit was not specified prior to the IETF standardization
effort. However, as far as was known to the working group, no FLAC
encoder at that time produced FLAC files containing residual samples
exceeding this limit. This is mostly because it is very unlikely to
encounter residual samples exceeding this limit when encoding 24-bit
PCM, and encoding of PCM with higher bit depths was not yet
implemented in any known encoder. In fact, these FLAC encoders would
produce corrupt files upon being triggered to produce such residual
samples, and it is unlikely any non-experimental encoder would ever
do so, even when presented with crafted material. Therefore, it was
not expected that existing implementations would be rendered non-
compliant by this change.
B.3. Addition of 5-Bit Rice Parameters
One significant addition to the format was the residual coding method
using 5-bit Rice parameters. Prior to publication of this addition
in July 2007, a partitioned Rice code with 4-bit Rice parameters was
the only residual coding method specified. The range offered by this
coding method proved too small when encoding 24-bit PCM; therefore, a
second residual coding method was specified that was identical to the
first, but with 5-bit Rice parameters.
B.4. Restriction of LPC Shift to Non-negative Values
As stated in Section 9.2.6, the predictor right shift is a number
signed two's complement, which MUST NOT be negative. This is because
shifting a number to the right by a negative amount is undefined
behavior in the C programming language standard. The intended
behavior was that a positive number would be a right shift and a
negative number would be a left shift. The FLAC reference encoder
was changed in 2007 to not generate LPC subframes with a negative
predictor right shift, as it turned out that the use of such
subframes would only very rarely provide any benefit and the decoders
that were already widely in use at that point were not able to handle
such subframes.
Appendix C. Interoperability Considerations
As documented in Appendix B, there have been some changes and
additions to the FLAC format. Additionally, implementation of
certain features of the FLAC format took many years, meaning early
decoder implementations could not be tested against files with these
features. Finally, many lower-quality FLAC decoders only implement
just enough features required for playback of the most common FLAC
files.
This appendix provides some considerations for encoder
implementations aiming to create highly compatible files. As this
topic is one that might change after this document is published,
consult [FLAC-wiki-interoperability] for more up-to-date information.
C.1. Features outside of the Streamable Subset
As described in Section 7, FLAC specifies a subset of its
capabilities as the FLAC streamable subset. Certain decoders may
choose to only decode FLAC files conforming to the limitations
imposed by the streamable subset. Therefore, maximum compatibility
with decoders is achieved when the limitations of the FLAC streamable
subset are followed when creating FLAC files.
C.2. Variable Block Size
Because it is often difficult to find the optimal arrangement of
block sizes for maximum compression, most encoders choose to create
files with a fixed block size. Because of this, many decoder
implementations receive minimal use when handling variable block size
streams, and this can reveal bugs or reveal that implementations do
not decode them at all. Furthermore, as explained in Appendix B.1,
there have been some changes to the way variable block size streams
are encoded. Because of this, maximum compatibility with decoders is
achieved when FLAC files are created using fixed block size streams.
C.3. 5-Bit Rice Parameters
As the addition of the coding method using 5-bit Rice parameters, as
described in Appendix B.3, occurred quite a few years after the FLAC
format was first introduced, some early decoders might not be able to
decode files containing such Rice parameters. The introduction of
this was specifically aimed at improving compression of 24-bit PCM
audio, and compression of 16-bit PCM audio only rarely benefits from
using 5-bit Rice parameters. Therefore, maximum compatibility with
decoders is achieved when FLAC files containing audio with a bit
depth of 16 bits or less are created without any use of 5-bit Rice
parameters.
C.4. Rice Escape Code
Escaped Rice partitions are seldom used, as it turned out their use
provides only a very small compression improvement. As many encoders
do not use these by default or are not capable of producing them at
all, it is likely that many decoder implementations are not able to
decode them correctly. Therefore, maximum compatibility with
decoders is achieved when FLAC files are created without any use of
escaped Rice partitions.
C.5. Uncommon Block Size
For unknown reasons, some decoders have chosen to support only common
block sizes for all but the last block of a stream. Therefore,
maximum compatibility with decoders is achieved when creating FLAC
files using common block sizes, as listed in Section 9.1.1, for all
but the last block of a stream.
C.6. Uncommon Bit Depth
Most audio is stored in bit depths that are a whole number of bytes,
e.g., 8, 16, or 24 bits. However, there is audio with different bit
depths. A few examples:
* DVD-Audio has the possibility to store 20-bit PCM audio.
* DAT and DV can store 12-bit PCM audio.
* NICAM-728 samples at 14 bits, which is companded to 10 bits.
* 8-bit µ-law can be losslessly converted to 14-bit (Linear) PCM.
* 8-bit A-law can be losslessly converted to 13-bit (Linear) PCM.
The FLAC format can contain these bit depths directly, but because
they are uncommon, some decoders are not able to process the
resulting files correctly. It is possible to store these formats in
a FLAC file with a more common bit depth without sacrificing
compression by padding each sample with zero bits to a bit depth that
is a whole byte. The FLAC format can efficiently compress these
wasted bits. See Section 9.2.2 for details.
Therefore, maximum compatibility with decoders is achieved when FLAC
files are created by padding samples of such audio with zero bits to
the bit depth that is the next whole number of bytes.
In cases where the original signal is already padded, this operation
cannot be reversed losslessly without knowing the original bit depth.
To leave no ambiguity, the original bit depth needs to be stored, for
example, in a Vorbis comment field or by storing the header of the
original file. The choice of a suitable method is left to the
implementor.
Besides audio with a "non-whole byte" bit depth, some decoder
implementations have chosen to only accept FLAC files coding for PCM
audio with a bit depth of 16 bits. Many implementations support bit
depths up to 24 bits, but no higher. Consult
[FLAC-wiki-interoperability] for more up-to-date information.
C.7. Multi-Channel Audio and Uncommon Sample Rates
Many FLAC audio players are unable to render multi-channel audio or
audio with an uncommon sample rate. While this is not a concern
specific to the FLAC format, it is of note when requiring maximum
compatibility with decoders. Unlike the previously mentioned
interoperability considerations, this is one where compatibility
cannot be improved without sacrificing the lossless nature of the
FLAC format.
From a non-exhaustive inquiry, it seems that a non-negligible number
of players, especially hardware players, do not support audio with 3
or more channels or sample rates other than those considered common;
see Section 9.1.2.
For those players that do support and are able to render multi-
channel audio, many do not parse and use the
WAVEFORMATEXTENSIBLE_CHANNEL_MASK tag (see Section 8.6.2). This is
also an interoperability consideration because compatibility cannot
be improved without sacrificing the lossless nature of the FLAC
format.
C.8. Changing Audio Properties Mid-Stream
Each FLAC frame header stores the audio sample rate, number of bits
per sample, and number of channels independently of the streaminfo
metadata block and other frame headers. This was done to permit
multicasting of FLAC files, but it also allows these properties to
change mid-stream. However, many FLAC decoders do not handle such
changes, as few other formats are capable of holding such streams and
changing playback properties during playback is often not possible
without interrupting playback. Also, as explained in Section 9,
using this feature of FLAC results in various practical problems.
However, even when storing an audio stream with changing properties
in FLAC encapsulated in a container capable of handling such changes,
as recommended in Section 9, many decoders are not able to decode
such a stream correctly. Therefore, maximum compatibility with
decoders is achieved when FLAC files are created with a single set of
audio properties, in which the properties coded in the streaminfo
metadata block (see Section 8.2) and the properties coded in all
frame headers (see Section 9.1) are the same. This can be achieved
by splitting up an input stream with changing audio properties at the
points where these properties change into separate streams or files.
Appendix D. Examples
This informational appendix contains short examples of FLAC files
that are decoded step by step. These examples provide a more
engaging way to understand the FLAC format than the formal
specification. The text explaining these examples assumes the reader
has at least cursorily read the specification and that the reader
refers to the specification for explanation of the terminology used.
These examples mostly focus on the layout of several metadata blocks,
subframe types, and the implications of certain aspects (e.g., wasted
bits and stereo decorrelation) on this layout.
The examples feature files generated by various FLAC encoders. These
are presented in hexadecimal or binary format, followed by tables and
text referring to various features by their starting bit positions in
these representations. Each starting position (shortened to "start"
in the tables) is a hexadecimal byte position and a start bit within
that byte, separated by a plus sign. Counts for these start at zero.
For example, a feature starting at the 3rd bit of the 17th byte is
referred to as starting at 0x10+2. The files that are explored in
these examples can be found at [FLAC-specification-github].
All data in this appendix has been thoroughly verified. However, as
this appendix is informational, if any information here conflicts
with statements in the formal specification, the latter takes
precedence.
D.1. Decoding Example 1
This very short example FLAC file codes for PCM audio that has two
channels, each containing one sample. The focus of this example is
on the essential parts of a FLAC file.
D.1.1. Example File 1 in Hexadecimal Representation
00000000: 664c 6143 8000 0022 1000 1000 fLaC..."....
0000000c: 0000 0f00 000f 0ac4 42f0 0000 ........B...
00000018: 0001 3e84 b418 07dc 6903 0758 ..>.....i..X
00000024: 6a3d ad1a 2e0f fff8 6918 0000 j=......i...
00000030: bf03 58fd 0312 8baa 9a ..X......
D.1.2. Example File 1 in Binary Representation
00000000: 01100110 01001100 01100001 01000011 fLaC
00000004: 10000000 00000000 00000000 00100010 ..."
00000008: 00010000 00000000 00010000 00000000 ....
0000000c: 00000000 00000000 00001111 00000000 ....
00000010: 00000000 00001111 00001010 11000100 ....
00000014: 01000010 11110000 00000000 00000000 B...
00000018: 00000000 00000001 00111110 10000100 ..>.
0000001c: 10110100 00011000 00000111 11011100 ....
00000020: 01101001 00000011 00000111 01011000 i..X
00000024: 01101010 00111101 10101101 00011010 j=..
00000028: 00101110 00001111 11111111 11111000 ....
0000002c: 01101001 00011000 00000000 00000000 i...
00000030: 10111111 00000011 01011000 11111101 ..X.
00000034: 00000011 00010010 10001011 10101010 ....
00000038: 10011010
D.1.3. Signature and Streaminfo
The first 4 bytes of the file contain the fLaC file signature.
Directly following it is a metadata block. The signature and the
first metadata block header are broken down in the following table.
+========+=========+============+===========================+
| Start | Length | Contents | Description |
+========+=========+============+===========================+
| 0x00+0 | 4 bytes | 0x664C6143 | fLaC |
+--------+---------+------------+---------------------------+
| 0x04+0 | 1 bit | 0b1 | Last metadata block |
+--------+---------+------------+---------------------------+
| 0x04+1 | 7 bits | 0b0000000 | Streaminfo metadata block |
+--------+---------+------------+---------------------------+
| 0x05+0 | 3 bytes | 0x000022 | Length of 34 bytes |
+--------+---------+------------+---------------------------+
Table 27
As the header indicates that this is the last metadata block, the
position of the first audio frame can now be calculated as the
position of the first byte after the metadata block header + the
length of the block, i.e., 8+34 = 42 or 0x2a. Thus, 0x2a indeed
contains the frame sync code for fixed block size streams -- 0xfff8.
The streaminfo metadata block contents are broken down in the
following table.
+========+==========+====================+==========================+
| Start | Length | Contents | Description |
+========+==========+====================+==========================+
| 0x08+0 | 2 bytes | 0x1000 | Min. block size 4096 |
+--------+----------+--------------------+--------------------------+
| 0x0a+0 | 2 bytes | 0x1000 | Max. block size 4096 |
+--------+----------+--------------------+--------------------------+
| 0x0c+0 | 3 bytes | 0x00000f | Min. frame size 15 bytes |
+--------+----------+--------------------+--------------------------+
| 0x0f+0 | 3 bytes | 0x00000f | Max. frame size 15 bytes |
+--------+----------+--------------------+--------------------------+
| 0x12+0 | 20 bits | 0x0ac4, 0b0100 | Sample rate 44100 hertz |
+--------+----------+--------------------+--------------------------+
| 0x14+4 | 3 bits | 0b001 | 2 channels |
+--------+----------+--------------------+--------------------------+
| 0x14+7 | 5 bits | 0b01111 | Sample bit depth 16 |
+--------+----------+--------------------+--------------------------+
| 0x15+4 | 36 bits | 0b0000, 0x00000001 | Total no. of samples 1 |
+--------+----------+--------------------+--------------------------+
| 0x1a | 16 | (...) | MD5 checksum |
| | bytes | | |
+--------+----------+--------------------+--------------------------+
Table 28
The minimum and maximum block sizes are both 4096. This was
apparently the block size the encoder planned to use, but as only 1
interchannel sample was provided, no frames with 4096 samples are
actually present in this file.
Note that anywhere a number of samples is mentioned (block size,
total number of samples, sample rate), interchannel samples are
meant.
The MD5 checksum (starting at 0x1a) is 0x3e84 b418 07dc 6903 0758
6a3d ad1a 2e0f. This will be validated after decoding the samples.
D.1.4. Audio Frames
The frame header starts at position 0x2a and is broken down in the
following table.
+========+=========+=================+===================+
| Start | Length | Contents | Description |
+========+=========+=================+===================+
| 0x2a+0 | 15 bits | 0xff, 0b1111100 | Frame sync |
+--------+---------+-----------------+-------------------+
| 0x2b+7 | 1 bit | 0b0 | Blocking strategy |
+--------+---------+-----------------+-------------------+
| 0x2c+0 | 4 bits | 0b0110 | 8-bit block size |
| | | | further down |
+--------+---------+-----------------+-------------------+
| 0x2c+4 | 4 bits | 0b1001 | Sample rate 44.1 |
| | | | kHz |
+--------+---------+-----------------+-------------------+
| 0x2d+0 | 4 bits | 0b0001 | Stereo, no |
| | | | decorrelation |
+--------+---------+-----------------+-------------------+
| 0x2d+4 | 3 bits | 0b100 | Bit depth 16 bits |
+--------+---------+-----------------+-------------------+
| 0x2d+7 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+-----------------+-------------------+
| 0x2e+0 | 1 byte | 0x00 | Frame number 0 |
+--------+---------+-----------------+-------------------+
| 0x2f+0 | 1 byte | 0x00 | Block size 1 |
+--------+---------+-----------------+-------------------+
| 0x30+0 | 1 byte | 0xbf | Frame header CRC |
+--------+---------+-----------------+-------------------+
Table 29
As the stream is a fixed block size stream, the number at 0x2e
contains a frame number. Because the value is smaller than 128, only
1 byte is used for the encoding.
At byte 0x31, the first subframe starts, which is broken down in the
following table.
+========+=========+================+=========================+
| Start | Length | Contents | Description |
+========+=========+================+=========================+
| 0x31+0 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+----------------+-------------------------+
| 0x31+1 | 6 bits | 0b000001 | Verbatim subframe |
+--------+---------+----------------+-------------------------+
| 0x31+7 | 1 bit | 0b1 | Wasted bits used |
+--------+---------+----------------+-------------------------+
| 0x32+0 | 2 bits | 0b01 | 2 wasted bits used |
+--------+---------+----------------+-------------------------+
| 0x32+2 | 14 bits | 0b011000, 0xfd | 14-bit unencoded sample |
+--------+---------+----------------+-------------------------+
Table 30
As the wasted bits flag is 1 in this subframe, a unary-coded number
follows. Starting at 0x32, we see 0b01, which unary codes for 1,
meaning that this subframe uses 2 wasted bits.
As this is a verbatim subframe, the subframe only contains unencoded
sample values. With a block size of 1, it contains only a single
sample. The bit depth of the audio is 16 bits, but as the subframe
header signals the use of 2 wasted bits, only 14 bits are stored. As
no stereo decorrelation is used, a bit depth increase for the side
channel is not applicable. So, the next 14 bits (starting at
position 0x32+2) contain the unencoded sample coded big-endian,
signed two's complement. The value reads 0b011000 11111101, or 6397.
This value needs to be shifted left by 2 bits to account for the
wasted bits. The value is then 0b011000 11111101 00, or 25588.
The second subframe starts at 0x34 and is broken down in the
following table.
+========+=========+==============+=========================+
| Start | Length | Contents | Description |
+========+=========+==============+=========================+
| 0x34+0 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+--------------+-------------------------+
| 0x34+1 | 6 bits | 0b000001 | Verbatim subframe |
+--------+---------+--------------+-------------------------+
| 0x34+7 | 1 bit | 0b1 | Wasted bits used |
+--------+---------+--------------+-------------------------+
| 0x35+0 | 4 bits | 0b0001 | 4 wasted bits used |
+--------+---------+--------------+-------------------------+
| 0x35+4 | 12 bits | 0b0010, 0x8b | 12-bit unencoded sample |
+--------+---------+--------------+-------------------------+
Table 31
The wasted bits flag is also one, but the unary-coded number that
follows it is 4 bits long, indicating the use of 4 wasted bits. This
means the sample is stored in 12 bits. The sample value is 0b0010
10001011, or 651. This value now has to be shifted left by 4 bits,
i.e., 0b0010 10001011 0000, or 10416.
At this point, we would undo stereo decorrelation if that was
applicable.
As the last subframe ends byte-aligned, no padding bits follow it.
The next 2 bytes, starting at 0x38, contain the frame CRC. As this
is the only frame in the file, the file ends with the CRC.
To validate the MD5 checksum, we line up the samples interleaved,
byte-aligned, little-endian, signed two's complement. The first
sample, with value 25588, translates to 0xf463, and the second
sample, with value 10416, translates to 0xb028. When computing the
MD5 checksum with 0xf463b028 as input, we get the MD5 checksum found
in the header, so decoding was lossless.
D.2. Decoding Example 2
This FLAC file is larger than the first example, but still contains
very little audio. The focus of this example is on decoding a
subframe with a fixed predictor and a coded residual, but it also
contains a very short seek table, a Vorbis comment metadata block,
and a padding metadata block.
D.2.1. Example File 2 in Hexadecimal Representation
00000000: 664c 6143 0000 0022 0010 0010 fLaC..."....
0000000c: 0000 1700 0044 0ac4 42f0 0000 .....D..B...
00000018: 0013 d5b0 5649 75e9 8b8d 8b93 ....VIu.....
00000024: 0422 757b 8103 0300 0012 0000 ."u{........
00000030: 0000 0000 0000 0000 0000 0000 ............
0000003c: 0000 0010 0400 003a 2000 0000 .......: ...
00000048: 7265 6665 7265 6e63 6520 6c69 reference li
00000054: 6246 4c41 4320 312e 332e 3320 bFLAC 1.3.3
00000060: 3230 3139 3038 3034 0100 0000 20190804....
0000006c: 0e00 0000 5449 544c 453d d7a9 ....TITLE=..
00000078: d79c d795 d79d 8100 0006 0000 ............
00000084: 0000 0000 fff8 6998 000f 9912 ......i.....
00000090: 0867 0162 3d14 4299 8f5d f70d .g.b=.B..]..
0000009c: 6fe0 0c17 caeb 2100 0ee7 a77a o.....!....z
000000a8: 24a1 590c 1217 b603 097b 784f $.Y......{xO
000000b4: aa9a 33d2 85e0 70ad 5b1b 4851 ..3...p.[.HQ
000000c0: b401 0d99 d2cd 1a68 f1e6 b810 .......h....
000000cc: fff8 6918 0102 a402 c382 c40b ..i.........
000000d8: c14a 03ee 48dd 03b6 7c13 30 .J..H...|.0
D.2.2. Example File 2 in Binary Representation (Only Audio Frames)
00000088: 11111111 11111000 01101001 10011000 ..i.
0000008c: 00000000 00001111 10011001 00010010 ....
00000090: 00001000 01100111 00000001 01100010 .g.b
00000094: 00111101 00010100 01000010 10011001 =.B.
00000098: 10001111 01011101 11110111 00001101 .]..
0000009c: 01101111 11100000 00001100 00010111 o...
000000a0: 11001010 11101011 00100001 00000000 ..!.
000000a4: 00001110 11100111 10100111 01111010 ...z
000000a8: 00100100 10100001 01011001 00001100 $.Y.
000000ac: 00010010 00010111 10110110 00000011 ....
000000b0: 00001001 01111011 01111000 01001111 .{xO
000000b4: 10101010 10011010 00110011 11010010 ..3.
000000b8: 10000101 11100000 01110000 10101101 ..p.
000000bc: 01011011 00011011 01001000 01010001 [.HQ
000000c0: 10110100 00000001 00001101 10011001 ....
000000c4: 11010010 11001101 00011010 01101000 ...h
000000c8: 11110001 11100110 10111000 00010000 ....
000000cc: 11111111 11111000 01101001 00011000 ..i.
000000d0: 00000001 00000010 10100100 00000010 ....
000000d4: 11000011 10000010 11000100 00001011 ....
000000d8: 11000001 01001010 00000011 11101110 .J..
000000dc: 01001000 11011101 00000011 10110110 H...
000000e0: 01111100 00010011 00110000 |.0
D.2.3. Streaminfo Metadata Block
Most of the streaminfo metadata block, including its header, is the
same as in example 1, so only parts that are different are listed in
the following table.
+========+=========+============+=============================+
| Start | Length | Contents | Description |
+========+=========+============+=============================+
| 0x04+0 | 1 bit | 0b0 | Not the last metadata block |
+--------+---------+------------+-----------------------------+
| 0x08+0 | 2 bytes | 0x0010 | Min. block size 16 |
+--------+---------+------------+-----------------------------+
| 0x0a+0 | 2 bytes | 0x0010 | Max. block size 16 |
+--------+---------+------------+-----------------------------+
| 0x0c+0 | 3 bytes | 0x000017 | Min. frame size 23 bytes |
+--------+---------+------------+-----------------------------+
| 0x0f+0 | 3 bytes | 0x000044 | Max. frame size 68 bytes |
+--------+---------+------------+-----------------------------+
| 0x15+4 | 36 bits | 0b0000, | Total no. of samples 19 |
| | | 0x00000013 | |
+--------+---------+------------+-----------------------------+
| 0x1a | 16 | (...) | MD5 checksum |
| | bytes | | |
+--------+---------+------------+-----------------------------+
Table 32
This time, the minimum and maximum block sizes are reflected in the
file: there is one block of 16 samples, and the last block (which has
3 samples) is not considered for the minimum block size. The MD5
checksum is 0xd5b0 5649 75e9 8b8d 8b93 0422 757b 8103. This will be
verified at the end of this example.
D.2.4. Seek Table
The seek table metadata block only holds one entry. It is not really
useful here, as it points to the first frame, but it is enough for
this example. The seek table metadata block is broken down in the
following table.
+========+========+====================+================+
| Start | Length | Contents | Description |
+========+========+====================+================+
| 0x2a+0 | 1 bit | 0b0 | Not the last |
| | | | metadata block |
+--------+--------+--------------------+----------------+
| 0x2a+1 | 7 bits | 0b0000011 | Seek table |
| | | | metadata block |
+--------+--------+--------------------+----------------+
| 0x2b+0 | 3 | 0x000012 | Length 18 |
| | bytes | | bytes |
+--------+--------+--------------------+----------------+
| 0x2e+0 | 8 | 0x0000000000000000 | Seek point to |
| | bytes | | sample 0 |
+--------+--------+--------------------+----------------+
| 0x36+0 | 8 | 0x0000000000000000 | Seek point to |
| | bytes | | offset 0 |
+--------+--------+--------------------+----------------+
| 0x3e+0 | 2 | 0x0010 | Seek point to |
| | bytes | | block size 16 |
+--------+--------+--------------------+----------------+
Table 33
D.2.5. Vorbis Comment
The Vorbis comment metadata block contains the vendor string and a
single comment. It is broken down in the following table.
+========+==========+============+===============================+
| Start | Length | Contents | Description |
+========+==========+============+===============================+
| 0x40+0 | 1 bit | 0b0 | Not the last metadata block |
+--------+----------+------------+-------------------------------+
| 0x40+1 | 7 bits | 0b0000100 | Vorbis comment metadata block |
+--------+----------+------------+-------------------------------+
| 0x41+0 | 3 bytes | 0x00003a | Length 58 bytes |
+--------+----------+------------+-------------------------------+
| 0x44+0 | 4 bytes | 0x20000000 | Vendor string length 32 bytes |
+--------+----------+------------+-------------------------------+
| 0x48+0 | 32 bytes | (...) | Vendor string |
+--------+----------+------------+-------------------------------+
| 0x68+0 | 4 bytes | 0x01000000 | Number of fields 1 |
+--------+----------+------------+-------------------------------+
| 0x6c+0 | 4 bytes | 0x0e000000 | Field length 14 bytes |
+--------+----------+------------+-------------------------------+
| 0x70+0 | 14 bytes | (...) | Field contents |
+--------+----------+------------+-------------------------------+
Table 34
The vendor string is reference libFLAC 1.3.3 20190804, and the field
contents of the only field is
TITLE=שלום
where in direction of reading, the sequence of characters forming the
field content is: "ש" (HEBREW LETTER SHIN, U+05E9), "ל" (HEBREW
LETTER LAMED, U+05DC), "ו" (HEBREW LETTER VAV, U+05D5), "ם" (HEBREW
LETTER FINAL MEM, U+05DD).
The Vorbis comment field is 14 bytes but only 10 characters in size,
because it contains four 2-byte characters.
D.2.6. Padding
The last metadata block is a (very short) padding block.
+========+=========+================+========================+
| Start | Length | Contents | Description |
+========+=========+================+========================+
| 0x7e+0 | 1 bit | 0b1 | Last metadata block |
+--------+---------+----------------+------------------------+
| 0x7e+1 | 7 bits | 0b0000001 | Padding metadata block |
+--------+---------+----------------+------------------------+
| 0x7f+0 | 3 bytes | 0x000006 | Length 6 byte |
+--------+---------+----------------+------------------------+
| 0x82+0 | 6 bytes | 0x000000000000 | Padding bytes |
+--------+---------+----------------+------------------------+
Table 35
D.2.7. First Audio Frame
The frame header starts at position 0x88 and is broken down in the
following table.
+========+=========+=================+===================+
| Start | Length | Contents | Description |
+========+=========+=================+===================+
| 0x88+0 | 15 bits | 0xff, 0b1111100 | Frame sync |
+--------+---------+-----------------+-------------------+
| 0x89+7 | 1 bit | 0b0 | Blocking strategy |
+--------+---------+-----------------+-------------------+
| 0x8a+0 | 4 bits | 0b0110 | 8-bit block size |
| | | | further down |
+--------+---------+-----------------+-------------------+
| 0x8a+4 | 4 bits | 0b1001 | Sample rate 44.1 |
| | | | kHz |
+--------+---------+-----------------+-------------------+
| 0x8b+0 | 4 bits | 0b1001 | Side-right stereo |
+--------+---------+-----------------+-------------------+
| 0x8b+4 | 3 bits | 0b100 | Bit depth 16 bit |
+--------+---------+-----------------+-------------------+
| 0x8b+7 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+-----------------+-------------------+
| 0x8c+0 | 1 byte | 0x00 | Frame number 0 |
+--------+---------+-----------------+-------------------+
| 0x8d+0 | 1 byte | 0x0f | Block size 16 |
+--------+---------+-----------------+-------------------+
| 0x8e+0 | 1 byte | 0x99 | Frame header CRC |
+--------+---------+-----------------+-------------------+
Table 36
The first subframe starts at byte 0x8f, and it is broken down in the
following table, excluding the coded residual. As this subframe
codes for a side channel, the bit depth is increased by 1 bit from 16
bits to 17 bits. This is most clearly present in the unencoded warm-
up sample.
+========+=========+=============+===========================+
| Start | Length | Contents | Description |
+========+=========+=============+===========================+
| 0x8f+0 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+-------------+---------------------------+
| 0x8f+1 | 6 bits | 0b001001 | Fixed subframe, 1st order |
+--------+---------+-------------+---------------------------+
| 0x8f+7 | 1 bit | 0b0 | No wasted bits used |
+--------+---------+-------------+---------------------------+
| 0x90+0 | 17 bits | 0x0867, 0b0 | Unencoded warm-up sample |
+--------+---------+-------------+---------------------------+
Table 37
The coded residual is broken down in the following table. All
quotients are unary coded, and all remainders are stored unencoded
with a number of bits specified by the Rice parameter.
+========+========+=================+=================+
| Start | Length | Contents | Description |
+========+========+=================+=================+
| 0x92+1 | 2 bits | 0b00 | Rice code with |
| | | | 4-bit parameter |
+--------+--------+-----------------+-----------------+
| 0x92+3 | 4 bits | 0b0000 | Partition order |
| | | | 0 |
+--------+--------+-----------------+-----------------+
| 0x92+7 | 4 bits | 0b1011 | Rice parameter |
| | | | 11 |
+--------+--------+-----------------+-----------------+
| 0x93+3 | 4 bits | 0b0001 | Quotient 3 |
+--------+--------+-----------------+-----------------+
| 0x93+7 | 11 | 0b00011110100 | Remainder 244 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0x95+2 | 2 bits | 0b01 | Quotient 1 |
+--------+--------+-----------------+-----------------+
| 0x95+4 | 11 | 0b01000100001 | Remainder 545 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0x96+7 | 2 bits | 0b01 | Quotient 1 |
+--------+--------+-----------------+-----------------+
| 0x97+1 | 11 | 0b00110011000 | Remainder 408 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0x98+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x98+5 | 11 | 0b11101011101 | Remainder 1885 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0x9a+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x9a+1 | 11 | 0b11101110000 | Remainder 1904 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0x9b+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x9b+5 | 11 | 0b10101101111 | Remainder 1391 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0x9d+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x9d+1 | 11 | 0b11000000000 | Remainder 1536 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0x9e+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x9e+5 | 11 | 0b10000010111 | Remainder 1047 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xa0+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa0+1 | 11 | 0b10010101110 | Remainder 1198 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xa1+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa1+5 | 11 | 0b01100100001 | Remainder 801 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xa3+0 | 13 | 0b0000000000001 | Quotient 12 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xa4+5 | 11 | 0b11011100111 | Remainder 1767 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xa6+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa6+1 | 11 | 0b01001110111 | Remainder 631 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xa7+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa7+5 | 11 | 0b01000100100 | Remainder 548 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xa9+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa9+1 | 11 | 0b01000010101 | Remainder 533 |
| | bits | | |
+--------+--------+-----------------+-----------------+
| 0xaa+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xaa+5 | 11 | 0b00100001100 | Remainder 268 |
| | bits | | |
+--------+--------+-----------------+-----------------+
Table 38
At this point, the decoder should know it is done decoding the coded
residual, as it received 16 samples: 1 warm-up sample and 15 residual
samples. Each residual sample can be calculated from the quotient
and remainder and from undoing the zigzag encoding. For example, the
value of the first zigzag-encoded residual sample is 3 * 2^11 + 244 =
6388. As this is an even number, the zigzag encoding is undone by
dividing by 2; the residual sample value is 3194. This is done for
all residual samples in the next table.
+==========+===========+================+=======================+
| Quotient | Remainder | Zigzag Encoded | Residual Sample Value |
+==========+===========+================+=======================+
| 3 | 244 | 6388 | 3194 |
+----------+-----------+----------------+-----------------------+
| 1 | 545 | 2593 | -1297 |
+----------+-----------+----------------+-----------------------+
| 1 | 408 | 2456 | 1228 |
+----------+-----------+----------------+-----------------------+
| 0 | 1885 | 1885 | -943 |
+----------+-----------+----------------+-----------------------+
| 0 | 1904 | 1904 | 952 |
+----------+-----------+----------------+-----------------------+
| 0 | 1391 | 1391 | -696 |
+----------+-----------+----------------+-----------------------+
| 0 | 1536 | 1536 | 768 |
+----------+-----------+----------------+-----------------------+
| 0 | 1047 | 1047 | -524 |
+----------+-----------+----------------+-----------------------+
| 0 | 1198 | 1198 | 599 |
+----------+-----------+----------------+-----------------------+
| 0 | 801 | 801 | -401 |
+----------+-----------+----------------+-----------------------+
| 12 | 1767 | 26343 | -13172 |
+----------+-----------+----------------+-----------------------+
| 0 | 631 | 631 | -316 |
+----------+-----------+----------------+-----------------------+
| 0 | 548 | 548 | 274 |
+----------+-----------+----------------+-----------------------+
| 0 | 533 | 533 | -267 |
+----------+-----------+----------------+-----------------------+
| 0 | 268 | 268 | 134 |
+----------+-----------+----------------+-----------------------+
Table 39
In this case, using a Rice code is more efficient than storing values
unencoded. The Rice code (excluding the partition order and
parameter) is 199 bits in length. The largest residual value
(-13172) would need 15 bits to be stored unencoded, so storing all 15
samples with 15 bits results in a sequence with a length of 225 bits.
The next step is using the predictor and the residuals to restore the
sample values. As this subframe uses a fixed predictor with order 1,
the residual value is added to the value of the previous sample.
+===========+==============+
| Residual | Sample Value |
+===========+==============+
| (warm-up) | 4302 |
+-----------+--------------+
| 3194 | 7496 |
+-----------+--------------+
| -1297 | 6199 |
+-----------+--------------+
| 1228 | 7427 |
+-----------+--------------+
| -943 | 6484 |
+-----------+--------------+
| 952 | 7436 |
+-----------+--------------+
| -696 | 6740 |
+-----------+--------------+
| 768 | 7508 |
+-----------+--------------+
| -524 | 6984 |
+-----------+--------------+
| 599 | 7583 |
+-----------+--------------+
| -401 | 7182 |
+-----------+--------------+
| -13172 | -5990 |
+-----------+--------------+
| -316 | -6306 |
+-----------+--------------+
| 274 | -6032 |
+-----------+--------------+
| -267 | -6299 |
+-----------+--------------+
| 134 | -6165 |
+-----------+--------------+
Table 40
With this, the decoding of the first subframe is complete. The
decoding of the second subframe is very similar, as it also uses a
fixed predictor of order 1. This is left as an exercise for the
reader; the results are in the next table. The next step is undoing
stereo decorrelation, which is done in the following table. As the
stereo decorrelation is side-right, the samples in the right channel
come directly from the second subframe, while the samples in the left
channel are found by adding the values of both subframes for each
sample.
+============+============+========+=======+
| Subframe 1 | Subframe 2 | Left | Right |
+============+============+========+=======+
| 4302 | 6070 | 10372 | 6070 |
+------------+------------+--------+-------+
| 7496 | 10545 | 18041 | 10545 |
+------------+------------+--------+-------+
| 6199 | 8743 | 14942 | 8743 |
+------------+------------+--------+-------+
| 7427 | 10449 | 17876 | 10449 |
+------------+------------+--------+-------+
| 6484 | 9143 | 15627 | 9143 |
+------------+------------+--------+-------+
| 7436 | 10463 | 17899 | 10463 |
+------------+------------+--------+-------+
| 6740 | 9502 | 16242 | 9502 |
+------------+------------+--------+-------+
| 7508 | 10569 | 18077 | 10569 |
+------------+------------+--------+-------+
| 6984 | 9840 | 16824 | 9840 |
+------------+------------+--------+-------+
| 7583 | 10680 | 18263 | 10680 |
+------------+------------+--------+-------+
| 7182 | 10113 | 17295 | 10113 |
+------------+------------+--------+-------+
| -5990 | -8428 | -14418 | -8428 |
+------------+------------+--------+-------+
| -6306 | -8895 | -15201 | -8895 |
+------------+------------+--------+-------+
| -6032 | -8476 | -14508 | -8476 |
+------------+------------+--------+-------+
| -6299 | -8896 | -15195 | -8896 |
+------------+------------+--------+-------+
| -6165 | -8653 | -14818 | -8653 |
+------------+------------+--------+-------+
Table 41
As the second subframe ends byte-aligned, no padding bits follow it.
Finally, the last 2 bytes of the frame contain the frame CRC.
D.2.8. Second Audio Frame
The second audio frame is very similar to the frame decoded in the
first example, but this time, 3 samples (not 1) are present.
The frame header starts at position 0xcc and is broken down in the
following table.
+========+=========+=================+===================+
| Start | Length | Contents | Description |
+========+=========+=================+===================+
| 0xcc+0 | 15 bits | 0xff, 0b1111100 | Frame sync |
+--------+---------+-----------------+-------------------+
| 0xcd+7 | 1 bit | 0b0 | Blocking strategy |
+--------+---------+-----------------+-------------------+
| 0xce+0 | 4 bits | 0b0110 | 8-bit block size |
| | | | further down |
+--------+---------+-----------------+-------------------+
| 0xce+4 | 4 bits | 0b1001 | Sample rate 44.1 |
| | | | kHz |
+--------+---------+-----------------+-------------------+
| 0xcf+0 | 4 bits | 0b0001 | Stereo, no |
| | | | decorrelation |
+--------+---------+-----------------+-------------------+
| 0xcf+4 | 3 bits | 0b100 | Bit depth 16 bits |
+--------+---------+-----------------+-------------------+
| 0xcf+7 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+-----------------+-------------------+
| 0xd0+0 | 1 byte | 0x01 | Frame number 1 |
+--------+---------+-----------------+-------------------+
| 0xd1+0 | 1 byte | 0x02 | Block size 3 |
+--------+---------+-----------------+-------------------+
| 0xd2+0 | 1 byte | 0xa4 | Frame header CRC |
+--------+---------+-----------------+-------------------+
Table 42
The first subframe starts at 0xd3+0 and is broken down in the
following table.
+========+=========+==========+=========================+
| Start | Length | Contents | Description |
+========+=========+==========+=========================+
| 0xd3+0 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+----------+-------------------------+
| 0xd3+1 | 6 bits | 0b000001 | Verbatim subframe |
+--------+---------+----------+-------------------------+
| 0xd3+7 | 1 bit | 0b0 | No wasted bits used |
+--------+---------+----------+-------------------------+
| 0xd4+0 | 16 bits | 0xc382 | 16-bit unencoded sample |
+--------+---------+----------+-------------------------+
| 0xd6+0 | 16 bits | 0xc40b | 16-bit unencoded sample |
+--------+---------+----------+-------------------------+
| 0xd8+0 | 16 bits | 0xc14a | 16-bit unencoded sample |
+--------+---------+----------+-------------------------+
Table 43
The second subframe starts at 0xda+0 and is broken down in the
following table.
+========+=========+===================+=========================+
| Start | Length | Contents | Description |
+========+=========+===================+=========================+
| 0xda+0 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+-------------------+-------------------------+
| 0xda+1 | 6 bits | 0b000001 | Verbatim subframe |
+--------+---------+-------------------+-------------------------+
| 0xda+7 | 1 bit | 0b1 | Wasted bits used |
+--------+---------+-------------------+-------------------------+
| 0xdb+0 | 1 bit | 0b1 | 1 wasted bit used |
+--------+---------+-------------------+-------------------------+
| 0xdb+1 | 15 bits | 0b110111001001000 | 15-bit unencoded sample |
+--------+---------+-------------------+-------------------------+
| 0xdd+0 | 15 bits | 0b110111010000001 | 15-bit unencoded sample |
+--------+---------+-------------------+-------------------------+
| 0xde+7 | 15 bits | 0b110110110011111 | 15-bit unencoded sample |
+--------+---------+-------------------+-------------------------+
Table 44
As this subframe uses wasted bits, the 15-bit unencoded samples need
to be shifted left by 1 bit. For example, sample 1 is stored as
-4536 and becomes -9072 after shifting left 1 bit.
As the last subframe does not end on byte alignment, 2 padding bits
are added before the 2-byte frame CRC, which follows at 0xe1+0.
D.2.9. MD5 Checksum Verification
All samples in the file have been decoded, and we can now verify the
MD5 checksum. All sample values must be interleaved and stored
signed coded little-endian. The result of this follows in groups of
12 samples (i.e., 6 interchannel samples) per line.
0x8428 B617 7946 3129 5E3A 2722 D445 D128 0B3D B723 EB45 DF28
0x723f 1E25 9D46 4929 B841 7026 5747 B829 8F43 8127 AEC7 14DF
0x9FC4 41DD 54C7 E4DE A5C4 40DD 1EC6 33DE 82C3 90DC 0BC4 02DD
0x4AC1 3EDB
The MD5 checksum of this is indeed the same as the one found in the
streaminfo metadata block.
D.3. Decoding Example 3
This example is once again a very short FLAC file. The focus of this
example is on decoding a subframe with a linear predictor and a coded
residual with more than one partition.
D.3.1. Example File 3 in Hexadecimal Representation
00000000: 664c 6143 8000 0022 1000 1000 fLaC..."....
0000000c: 0000 1f00 001f 07d0 0070 0000 .........p..
00000018: 0018 f8f9 e396 f5cb cfc6 dc80 ............
00000024: 7f99 7790 6b32 fff8 6802 0017 ..w.k2..h...
00000030: e944 004f 6f31 3d10 47d2 27cb .D.Oo1=.G.'.
0000003c: 6d09 0831 452b dc28 2222 8057 m..1E+.("".W
00000048: a3 .
D.3.2. Example File 3 in Binary Representation (Only Audio Frame)
0000002a: 11111111 11111000 01101000 00000010 ..h.
0000002e: 00000000 00010111 11101001 01000100 ...D
00000032: 00000000 01001111 01101111 00110001 .Oo1
00000036: 00111101 00010000 01000111 11010010 =.G.
0000003a: 00100111 11001011 01101101 00001001 '.m.
0000003e: 00001000 00110001 01000101 00101011 .1E+
00000042: 11011100 00101000 00100010 00100010 .(""
00000046: 10000000 01010111 10100011 .W.
D.3.3. Streaminfo Metadata Block
Most of the streaminfo metadata block, including its header, is the
same as in example 1, so only parts that are different are listed in
the following table.
+========+==========+====================+==========================+
| Start | Length | Contents | Description |
+========+==========+====================+==========================+
| 0x0c+0 | 3 bytes | 0x00001f | Min. frame size 31 bytes |
+--------+----------+--------------------+--------------------------+
| 0x0f+0 | 3 bytes | 0x00001f | Max. frame size 31 bytes |
+--------+----------+--------------------+--------------------------+
| 0x12+0 | 20 bits | 0x07d0, 0x0000 | Sample rate 32000 hertz |
+--------+----------+--------------------+--------------------------+
| 0x14+4 | 3 bits | 0b000 | 1 channel |
+--------+----------+--------------------+--------------------------+
| 0x14+7 | 5 bits | 0b00111 | Sample bit depth 8 bits |
+--------+----------+--------------------+--------------------------+
| 0x15+4 | 36 bits | 0b0000, 0x00000018 | Total no. of samples 24 |
+--------+----------+--------------------+--------------------------+
| 0x1a | 16 | (...) | MD5 checksum |
| | bytes | | |
+--------+----------+--------------------+--------------------------+
Table 45
D.3.4. Audio Frame
The frame header starts at position 0x2a and is broken down in the
following table.
+========+=========+=================+===================+
| Start | Length | Contents | Description |
+========+=========+=================+===================+
| 0x2a+0 | 15 bits | 0xff, 0b1111100 | Frame sync |
+--------+---------+-----------------+-------------------+
| 0x2b+7 | 1 bit | 0b0 | blocking strategy |
+--------+---------+-----------------+-------------------+
| 0x2c+0 | 4 bits | 0b0110 | 8-bit block size |
| | | | further down |
+--------+---------+-----------------+-------------------+
| 0x2c+4 | 4 bits | 0b1000 | Sample rate 32 |
| | | | kHz |
+--------+---------+-----------------+-------------------+
| 0x2d+0 | 4 bits | 0b0000 | Mono audio (1 |
| | | | channel) |
+--------+---------+-----------------+-------------------+
| 0x2d+4 | 3 bits | 0b001 | Bit depth 8 bits |
+--------+---------+-----------------+-------------------+
| 0x2d+7 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+---------+-----------------+-------------------+
| 0x2e+0 | 1 byte | 0x00 | Frame number 0 |
+--------+---------+-----------------+-------------------+
| 0x2f+0 | 1 byte | 0x17 | Block size 24 |
+--------+---------+-----------------+-------------------+
| 0x30+0 | 1 byte | 0xe9 | Frame header CRC |
+--------+---------+-----------------+-------------------+
Table 46
The first and only subframe starts at byte 0x31. It is broken down
in the following table, without the coded residual.
+========+========+==========+=====================+
| Start | Length | Contents | Description |
+========+========+==========+=====================+
| 0x31+0 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+--------+----------+---------------------+
| 0x31+1 | 6 bits | 0b100010 | Linear prediction |
| | | | subframe, 3rd order |
+--------+--------+----------+---------------------+
| 0x31+7 | 1 bit | 0b0 | No wasted bits used |
+--------+--------+----------+---------------------+
| 0x32+0 | 8 bits | 0x00 | Unencoded warm-up |
| | | | sample 0 |
+--------+--------+----------+---------------------+
| 0x33+0 | 8 bits | 0x4f | Unencoded warm-up |
| | | | sample 79 |
+--------+--------+----------+---------------------+
| 0x34+0 | 8 bits | 0x6f | Unencoded warm-up |
| | | | sample 111 |
+--------+--------+----------+---------------------+
| 0x35+0 | 4 bits | 0b0011 | Coefficient |
| | | | precision 4 bit |
+--------+--------+----------+---------------------+
| 0x35+4 | 5 bits | 0b00010 | Prediction right |
| | | | shift 2 |
+--------+--------+----------+---------------------+
| 0x36+1 | 4 bits | 0b0111 | Predictor |
| | | | coefficient 7 |
+--------+--------+----------+---------------------+
| 0x36+5 | 4 bits | 0b1010 | Predictor |
| | | | coefficient -6 |
+--------+--------+----------+---------------------+
| 0x37+1 | 4 bits | 0b0010 | Predictor |
| | | | coefficient 2 |
+--------+--------+----------+---------------------+
Table 47
The data stream continues with the coded residual, which is broken
down in the following table. Residual partitions 3 and 4 are left as
an exercise for the reader.
+========+========+==========+======================================+
| Start | Length | Contents | Description |
+========+========+==========+======================================+
| 0x37+5 | 2 bits | 0b00 | Rice-coded residual, |
| | | | 4-bit parameter |
+--------+--------+----------+--------------------------------------+
| 0x37+7 | 4 bits | 0b0010 | Partition order 2 |
+--------+--------+----------+--------------------------------------+
| 0x38+3 | 4 bits | 0b0011 | Rice parameter 3 |
+--------+--------+----------+--------------------------------------+
| 0x38+7 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+----------+--------------------------------------+
| 0x39+0 | 3 bits | 0b110 | Remainder 6 |
+--------+--------+----------+--------------------------------------+
| 0x39+3 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+----------+--------------------------------------+
| 0x39+4 | 3 bits | 0b001 | Remainder 1 |
+--------+--------+----------+--------------------------------------+
| 0x39+7 | 4 bits | 0b0001 | Quotient 3 |
+--------+--------+----------+--------------------------------------+
| 0x3a+3 | 3 bits | 0b001 | Remainder 1 |
+--------+--------+----------+--------------------------------------+
| 0x3a+6 | 4 bits | 0b1111 | No Rice parameter, |
| | | | escape code |
+--------+--------+----------+--------------------------------------+
| 0x3b+2 | 5 bits | 0b00101 | Partition encoded |
| | | | with 5 bits |
+--------+--------+----------+--------------------------------------+
| 0x3b+7 | 5 bits | 0b10110 | Residual -10 |
+--------+--------+----------+--------------------------------------+
| 0x3c+4 | 5 bits | 0b11010 | Residual -6 |
+--------+--------+----------+--------------------------------------+
| 0x3d+1 | 5 bits | 0b00010 | Residual 2 |
+--------+--------+----------+--------------------------------------+
| 0x3d+6 | 5 bits | 0b01000 | Residual 8 |
+--------+--------+----------+--------------------------------------+
| 0x3e+3 | 5 bits | 0b01000 | Residual 8 |
+--------+--------+----------+--------------------------------------+
| 0x3f+0 | 5 bits | 0b00110 | Residual 6 |
+--------+--------+----------+--------------------------------------+
| 0x3f+5 | 4 bits | 0b0010 | Rice parameter 2 |
+--------+--------+----------+--------------------------------------+
| 0x40+1 | 22 | (...) | Residual partition 3 |
| | bits | | |
+--------+--------+----------+--------------------------------------+
| 0x42+7 | 4 bits | 0b0001 | Rice parameter 1 |
+--------+--------+----------+--------------------------------------+
| 0x43+3 | 23 | (...) | Residual partition 4 |
| | bits | | |
+--------+--------+----------+--------------------------------------+
Table 48
The frame ends with 6 padding bits and a 2-byte frame CRC.
To decode this subframe, 21 predictions have to be calculated and
added to their corresponding residuals. This is a sequential
process: as each prediction uses previous samples, it is not possible
to start this decoding halfway through a subframe or decode a
subframe with parallel threads.
The following table breaks down the calculation for each sample. For
example, the predictor without shift value of row 4 is found by
applying the predictor with the three warm-up samples: 7*111 - 6*79 +
2*0 = 303. This value is then shifted right by 2 bits: 303 >> 2 =
75. Then, the decoded residual sample is added: 75 + 3 = 78.
+===========+=====================+===========+==============+
| Residual | Predictor w/o Shift | Predictor | Sample Value |
+===========+=====================+===========+==============+
| (warm-up) | N/A | N/A | 0 |
+-----------+---------------------+-----------+--------------+
| (warm-up) | N/A | N/A | 79 |
+-----------+---------------------+-----------+--------------+
| (warm-up) | N/A | N/A | 111 |
+-----------+---------------------+-----------+--------------+
| 3 | 303 | 75 | 78 |
+-----------+---------------------+-----------+--------------+
| -1 | 38 | 9 | 8 |
+-----------+---------------------+-----------+--------------+
| -13 | -190 | -48 | -61 |
+-----------+---------------------+-----------+--------------+
| -10 | -319 | -80 | -90 |
+-----------+---------------------+-----------+--------------+
| -6 | -248 | -62 | -68 |
+-----------+---------------------+-----------+--------------+
| 2 | -58 | -15 | -13 |
+-----------+---------------------+-----------+--------------+
| 8 | 137 | 34 | 42 |
+-----------+---------------------+-----------+--------------+
| 8 | 236 | 59 | 67 |
+-----------+---------------------+-----------+--------------+
| 6 | 191 | 47 | 53 |
+-----------+---------------------+-----------+--------------+
| 0 | 53 | 13 | 13 |
+-----------+---------------------+-----------+--------------+
| -3 | -93 | -24 | -27 |
+-----------+---------------------+-----------+--------------+
| -5 | -161 | -41 | -46 |
+-----------+---------------------+-----------+--------------+
| -4 | -134 | -34 | -38 |
+-----------+---------------------+-----------+--------------+
| -1 | -44 | -11 | -12 |
+-----------+---------------------+-----------+--------------+
| 1 | 52 | 13 | 14 |
+-----------+---------------------+-----------+--------------+
| 1 | 94 | 23 | 24 |
+-----------+---------------------+-----------+--------------+
| 4 | 60 | 15 | 19 |
+-----------+---------------------+-----------+--------------+
| 2 | 17 | 4 | 6 |
+-----------+---------------------+-----------+--------------+
| 2 | -24 | -6 | -4 |
+-----------+---------------------+-----------+--------------+
| 2 | -26 | -7 | -5 |
+-----------+---------------------+-----------+--------------+
| 0 | 1 | 0 | 0 |
+-----------+---------------------+-----------+--------------+
Table 49
By lining up all these samples, we get the following input for the
MD5 checksum calculation process:
0x004F 6F4E 08C3 A6BC F32A 4335 0DE5 D2DA F40E 1813 06FC FB00
This indeed results in the MD5 checksum found in the streaminfo
metadata block.
Acknowledgments
FLAC owes much to the many people who have advanced the audio
compression field so freely. For instance:
* Tony Robinson: He worked on Shorten, and his paper (see
[Robinson-TR156]) 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.
* Solomon W. Golomb and Robert F. Rice: Their universal codes are
used by FLAC's entropy coder. See [Rice].
* Norman Levinson and James Durbin: The FLAC reference encoder uses
an algorithm developed and refined by them for determining the LPC
coefficients from the autocorrelation coefficients. See
[Durbin]).
* Claude Shannon: See [Shannon].
The FLAC format, the FLAC reference implementation
[FLAC-implementation], and the initial draft version of this document
were originally developed by Josh Coalson. While many others have
contributed since, this original effort is deeply appreciated.
Authors' Addresses
Martijn van Beurden
Netherlands
Email: mvanb1@gmail.com
Andrew Weaver
Email: theandrewjw@gmail.com
ERRATA