Internet DRAFT - draft-ietf-cellar-flac
draft-ietf-cellar-flac
cellar M.Q.C. van Beurden
Internet-Draft
Intended status: Standards Track A. Weaver
Expires: 5 October 2023 3 April 2023
Free Lossless Audio Codec
draft-ietf-cellar-flac-08
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 without
losing information in doing so (i.e. lossless). 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 coded to and from
with little computing resources. Decoding of FLAC has seen many
independent implementations on many different platforms, and both
encoding and decoding can be implemented without needing floating-
point arithmetic.
Status of This Memo
This Internet-Draft is submitted in full conformance with the
provisions of BCP 78 and BCP 79.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF). Note that other groups may also distribute
working documents as Internet-Drafts. The list of current Internet-
Drafts is at https://datatracker.ietf.org/drafts/current/.
Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any
time. It is inappropriate to use Internet-Drafts as reference
material or to cite them other than as "work in progress."
This Internet-Draft will expire on 5 October 2023.
Copyright Notice
Copyright (c) 2023 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.
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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 . . . . . . . . . . . . . . . . . . . . . . . . 4
2. Notation and Conventions . . . . . . . . . . . . . . . . . . 4
3. Definitions . . . . . . . . . . . . . . . . . . . . . . . . . 5
4. Conceptual overview . . . . . . . . . . . . . . . . . . . . . 6
4.1. Blocking . . . . . . . . . . . . . . . . . . . . . . . . 7
4.2. Interchannel Decorrelation . . . . . . . . . . . . . . . 8
4.3. Prediction . . . . . . . . . . . . . . . . . . . . . . . 9
4.4. Residual Coding . . . . . . . . . . . . . . . . . . . . . 10
5. Format principles . . . . . . . . . . . . . . . . . . . . . . 11
6. Format lay-out . . . . . . . . . . . . . . . . . . . . . . . 13
7. Streamable subset . . . . . . . . . . . . . . . . . . . . . . 14
8. File-level metadata . . . . . . . . . . . . . . . . . . . . . 15
8.1. Metadata block header . . . . . . . . . . . . . . . . . . 15
8.2. Streaminfo . . . . . . . . . . . . . . . . . . . . . . . 16
8.3. Padding . . . . . . . . . . . . . . . . . . . . . . . . . 19
8.4. Application . . . . . . . . . . . . . . . . . . . . . . . 19
8.5. Seektable . . . . . . . . . . . . . . . . . . . . . . . . 20
8.5.1. Seekpoint . . . . . . . . . . . . . . . . . . . . . . 21
8.6. Vorbis comment . . . . . . . . . . . . . . . . . . . . . 21
8.6.1. Standard field names . . . . . . . . . . . . . . . . 22
8.6.2. Channel mask . . . . . . . . . . . . . . . . . . . . 23
8.7. Cuesheet . . . . . . . . . . . . . . . . . . . . . . . . 25
8.7.1. Cuesheet track . . . . . . . . . . . . . . . . . . . 27
8.8. Picture . . . . . . . . . . . . . . . . . . . . . . . . . 28
9. Frame structure . . . . . . . . . . . . . . . . . . . . . . . 32
9.1. Frame header . . . . . . . . . . . . . . . . . . . . . . 33
9.1.1. Block size bits . . . . . . . . . . . . . . . . . . . 33
9.1.2. Sample rate bits . . . . . . . . . . . . . . . . . . 34
9.1.3. Channels bits . . . . . . . . . . . . . . . . . . . . 35
9.1.4. Bit depth bits . . . . . . . . . . . . . . . . . . . 36
9.1.5. Coded number . . . . . . . . . . . . . . . . . . . . 37
9.1.6. Uncommon block size . . . . . . . . . . . . . . . . . 39
9.1.7. Uncommon sample rate . . . . . . . . . . . . . . . . 39
9.1.8. Frame header CRC . . . . . . . . . . . . . . . . . . 40
9.2. Subframes . . . . . . . . . . . . . . . . . . . . . . . . 40
9.2.1. Subframe header . . . . . . . . . . . . . . . . . . . 40
9.2.2. Wasted bits per sample . . . . . . . . . . . . . . . 41
9.2.3. Constant subframe . . . . . . . . . . . . . . . . . . 42
9.2.4. Verbatim subframe . . . . . . . . . . . . . . . . . . 42
9.2.5. Fixed predictor subframe . . . . . . . . . . . . . . 42
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9.2.6. Linear predictor subframe . . . . . . . . . . . . . . 44
9.2.7. Coded residual . . . . . . . . . . . . . . . . . . . 45
9.3. Frame footer . . . . . . . . . . . . . . . . . . . . . . 48
10. Container mappings . . . . . . . . . . . . . . . . . . . . . 49
10.1. Ogg mapping . . . . . . . . . . . . . . . . . . . . . . 49
10.2. Matroska mapping . . . . . . . . . . . . . . . . . . . . 51
10.3. ISO Base Media File Format (MP4) mapping . . . . . . . . 52
11. Implementation status . . . . . . . . . . . . . . . . . . . . 52
12. Security Considerations . . . . . . . . . . . . . . . . . . . 53
13. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 55
13.1. Media type registration . . . . . . . . . . . . . . . . 55
14. Acknowledgments . . . . . . . . . . . . . . . . . . . . . . . 56
15. Normative References . . . . . . . . . . . . . . . . . . . . 57
16. Informative References . . . . . . . . . . . . . . . . . . . 57
Appendix A. Numerical considerations . . . . . . . . . . . . . . 59
A.1. Determining necessary data type size . . . . . . . . . . 59
A.2. Stereo decorrelation . . . . . . . . . . . . . . . . . . 60
A.3. Prediction . . . . . . . . . . . . . . . . . . . . . . . 60
A.4. Residual . . . . . . . . . . . . . . . . . . . . . . . . 62
A.5. Rice coding . . . . . . . . . . . . . . . . . . . . . . . 63
Appendix B. Past format changes . . . . . . . . . . . . . . . . 63
B.1. Addition of block size strategy flag . . . . . . . . . . 63
B.2. Restriction of encoded residual samples . . . . . . . . . 64
B.3. Addition of 5-bit Rice parameter . . . . . . . . . . . . 64
B.4. Restriction of LPC shift to non-negative values . . . . . 64
Appendix C. Interoperability considerations . . . . . . . . . . 65
C.1. Features outside of streamable subset . . . . . . . . . . 65
C.2. Variable block size . . . . . . . . . . . . . . . . . . . 65
C.3. 5-bit Rice parameter . . . . . . . . . . . . . . . . . . 66
C.4. Rice escape code . . . . . . . . . . . . . . . . . . . . 66
C.5. Uncommon block size . . . . . . . . . . . . . . . . . . . 66
C.6. Uncommon bit depth . . . . . . . . . . . . . . . . . . . 66
C.7. Multi-channel audio and uncommon sample rates . . . . . . 67
Appendix D. Examples . . . . . . . . . . . . . . . . . . . . . . 67
D.1. Decoding example 1 . . . . . . . . . . . . . . . . . . . 68
D.1.1. Example file 1 in hexadecimal representation . . . . 68
D.1.2. Example file 1 in binary representation . . . . . . . 68
D.1.3. Signature and streaminfo . . . . . . . . . . . . . . 69
D.1.4. Audio frames . . . . . . . . . . . . . . . . . . . . 70
D.2. Decoding example 2 . . . . . . . . . . . . . . . . . . . 73
D.2.1. Example file 2 in hexadecimal representation . . . . 73
D.2.2. Example file 2 in binary representation (only audio
frames) . . . . . . . . . . . . . . . . . . . . . . . 74
D.2.3. Streaminfo metadata block . . . . . . . . . . . . . . 75
D.2.4. Seektable . . . . . . . . . . . . . . . . . . . . . . 75
D.2.5. Vorbis comment . . . . . . . . . . . . . . . . . . . 76
D.2.6. Padding . . . . . . . . . . . . . . . . . . . . . . . 77
D.2.7. First audio frame . . . . . . . . . . . . . . . . . . 78
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D.2.8. Second audio frame . . . . . . . . . . . . . . . . . 84
D.2.9. MD5 checksum verification . . . . . . . . . . . . . . 86
D.3. Decoding example 3 . . . . . . . . . . . . . . . . . . . 86
D.3.1. Example file 3 in hexadecimal representation . . . . 86
D.3.2. Example file 3 in binary representation (only audio
frame) . . . . . . . . . . . . . . . . . . . . . . . 86
D.3.3. Streaminfo metadata block . . . . . . . . . . . . . . 86
D.3.4. Audio frame . . . . . . . . . . . . . . . . . . . . . 87
Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 92
1. Introduction
This document defines the 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 up to 1048575 hertz and bit
depths from 4 up 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 most 32 samples.
The coding methods provided by the FLAC format work best on PCM audio
signals of which the samples 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 to compress
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 unsigned big-endian integer using
n bits. Values expressed as s(n) represent 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
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bits expressed is represented with an ellipsis, such as u(m...n).
The name of a value followed by an asterisk * indicates zero or more
occurrences of the value. The name of a value followed by a plus
sign + indicates one or more occurrences of the value.
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 pulse-code modulated audio,
with one or more channels.
* *Subblock*: All samples within a corresponding block for 1
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. A subframe corresponds to a
subblock, the average of two subblocks or the difference between
two subblocks, see section on interchannel decorrelation
(#interchannel-decorrelation).
* *Block size*: The total number of samples contained in a block or
coded in a frame, divided by the number of channels. In other
words, the number of samples in any subblock of a block, or any
subframe of a frame. This is also called *interchannel samples*.
* *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 (#interchannel-
decorrelation).
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* *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 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 smaller
numerical value than the original signal.
* *Rice code*: A variable-length code (see [VarLengthCode]) which
compresses data by making use of the observation that, after using
an effective predictor, most residual samples are closer to zero
than the original samples, while still allowing for a small part
of the samples to be much larger.
4. Conceptual overview
Similar to many other audio coders, a FLAC file is encoded following
the steps below. On decoding a FLAC file, these steps are undone in
reverse order, i.e. from bottom to top.
* Blocking (see section on Blocking (#blocking)). The input is
split up into many contiguous blocks.
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* Interchannel Decorrelation (see section on Interchannel
Decorrelation (#interchannel-decorrelation)). In the case of
stereo streams, the FLAC format allows for transforming the left-
right signal into a mid-side signal to remove redundancy, if there
is any. Besides coding as left-right and mid-side, it is also
possible to code left-side and side-right, whichever ordering
results in the highest compression. Choosing between any of these
transformation is done independently for each block.
* Prediction (see section on Prediction (#prediction)). 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 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 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.
* Residual Coding (See section on Residual Coding (#residual-
coding)). 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 on File-
level metadata (#file-level-metadata)), which allows arbitrary
information about the stream to be included at the beginning of the
stream.
4.1. Blocking
The size used for blocking the audio data has a direct effect on the
compression ratio. If the block size is too small, the resulting
large number of frames mean that excess bits will be wasted on frame
headers. If the block size is too large, the characteristics of the
signal may vary so much that the encoder will be unable to find a
good predictor. In order to simplify encoder/decoder design, FLAC
imposes a minimum block size of 16 samples, 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
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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 the FLAC format explicitly
stores whether a file has a constant or a variable block size
throughout the stream, and stores a block number instead of a sample
number to slightly improve compression if a stream has a constant
block size.
Blocked data is passed to the predictor stage one subblock at a time.
Each subblock is independently coded into a subframe, and the
subframes are concatenated into a frame. Because each channel is
coded separately, subframes MAY use different predictors, even within
a frame.
4.2. Interchannel Decorrelation
In many audio files, channels are correlated. The FLAC format can
exploit this correlation in stereo files by not directly coding
subblocks into subframes, but instead coding an average of all
samples in both subblocks (a mid channel) or the difference between
all samples in both subblocks (a side channel). 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 one 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, while
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
subblock 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 the left subframe.
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* *Right-side*. The right subblock is coded and the left and right
subblock are used to code a side subframe. Note that the actual
coded subframe order is side-right. 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 one bit. The right shift
of the mid channel does not lead to non-lossless 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 like white
noise. 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, 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 the
section on the fixed predictor subframe (#fixed-predictor-
subframe)
4. *Linear predictor*. Samples are predicted using past samples and
a set of predictor coefficients, 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,
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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 the section on
the linear predictor subframe (#linear-predictor-subframe)
A FLAC encoder is free in selecting which method is used 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 of
which all residual samples are representable in 32-bit signed
integers as stated in section coded residual (#coded-residual), a
verbatim subframe is used.
For more information on fixed and linear predictors, see
[HPL-1999-144] 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 a
few 'outlier' residual samples are much larger than any of the
original samples. Sometimes these outliers even exceed the range the
bit depth of the original audio offers.
To be able 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 in two parts by dividing it with 2^(Rice
parameter), creating a quotient and a remainder. The quotient is
stored in unary form, the remainder in binary form. If indeed most
residual samples are close to zero and the Rice parameter is chosen
right, this form of coding, a so-called variable-length code, needs
less bits to store than storing the residual in unencoded form.
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As Rice codes can only handle unsigned numbers, signed numbers are
zigzag encoded to a so-called folded residual. For more information
see section coded residual (#coded-residual) 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 or skip data encoded
using methods they do not recognize. Apart from reserved patterns,
the format specifies invalid patterns in certain places, meaning that
the patterns MUST NOT appear in any bitstream. These invalid
patterns are usually used to make the synchronization mechanism more
robust. They are listed in the following table.
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+===========================+==============================+
| Description | Reference |
+===========================+==============================+
| Metadata block type 127 | Metadata block header |
| | (#metadata-block-header) |
+---------------------------+------------------------------+
| Minimum and maximum block | Streaminfo metadata block |
| sizes smaller than 16 in | (#streaminfo) |
| streaminfo metadata block | |
+---------------------------+------------------------------+
| Sample rate bits 0b1111 | Sample rate bits (#sample- |
| | rate-bits) |
+---------------------------+------------------------------+
| Uncommon blocksize 65536 | Uncommon block size |
| | (#uncommon-block-size) |
+---------------------------+------------------------------+
| Predictor coefficient | Linear predictor subframe |
| precision bits 0b1111 | (#linear-predictor-subframe) |
+---------------------------+------------------------------+
| Negative predictor right | Linear predictor subframe |
| shift | (#linear-predictor-subframe) |
+---------------------------+------------------------------+
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 length used in Vorbis comments (see Vorbis comment
metadata block (#vorbis-comment)), which are little-endian coded.
All numbers are unsigned except linear predictor coefficients, the
linear prediction shift (see linear predictor subframe (#linear-
predictor-subframe)) and numbers which 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 audio signals of which the numerical values
of the samples are centered around zero, i.e. have no DC offset. In
most unsigned audio formats, signals are centered around halfway the
range of the unsigned integer type used. If that is the case, all
sample representations SHOULD be converted by first copying the
number to a signed integer with sufficient range and then subtracting
half of the range of the unsigned integer type, which should result
in a signal with samples centered around 0.
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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 invalid data, decoder behavior is left
unspecified. A decoder MAY choose to stop decoding on encountering
such invalid data. Examples of such invalid data are
* One or more decoded sample values exceeds the range offered by the
bit depth as coded for that frame. E.g. in a frame with a bit
depth of 8 bits, any samples not in the inclusive range from -128
to 127 are invalid.
* The number of wasted bits (see section wasted bits per sample
(#wasted-bits-per-sample)) of a subframe is such that the bit
depth of that subframe (see section constant subframe (#constant-
subframe) for a description of subframe bit depth) equals zero or
is negative
* A frame header CRC (see section frame header CRC (#frame-header-
crc)) or frame footer CRC (see section frame footer (#frame-
footer)) does not validate
* One of the invalid bit patterns described in table 1 above is used
6. Format lay-out
Before the formal description of the stream, an overview of the lay-
out of FLAC file might be helpful.
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 block), any number of other metadata blocks,
then the audio frames.
FLAC supports up to 128 kinds of metadata blocks; currently 7 kinds
are defined in the section file-level metadata (#file-level-
metadata).
The audio data is composed of one or more audio frames. Each frame
consists of a frame header, which contains a sync code, information
about the frame like the block size, sample rate, number of channels,
et cetera, and an 8-bit CRC. The frame header also contains either
the sample number of the first sample in the frame (for variable
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.
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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 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 pure overhead, it has a
direct effect on the compression ratio. To keep the frame header as
small as possible, FLAC uses lookup tables for the most commonly used
values for frame 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 streamable subset (#streamable-subset) for
details. In this way, the file is streamable and the frame header
size small for all of the most common forms of audio data.
Individual subframes (one for each channel) are coded separately
within a frame, and appear serially in the stream. In other words,
the encoded audio data is NOT channel-interleaved. This reduces
decoder complexity at the cost of requiring larger decode buffers.
Each subframe has its own header specifying the attributes of the
subframe, like prediction method and order, residual coding
parameters, etc. 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 such that decoder buffer sizes and other resource
requirements can be easily determined. The flac command-line tool,
part of the FLAC reference implementation, (see section
implementation status (#implementation-status)) generates streamable
subset files by default unless the --lax command-line option is used.
The streamable subset makes the following limitations on what MAY be
used in the stream:
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* The sample rate bits (#sample-rate-bits) 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 bits depth bits (#bit-depth-bits) 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 inter-
channel 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 inter-channel samples,
i.e. the maximum block size used for this audio must not be larger
than 4608.
* Linear prediction subframes (see section linear predictor subframe
(#linear-predictor-subframe)) 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 subframe header section (#subframe-header)) MUST NOT
be 0b101100-0b111111.
* The Rice partition order (see coded residual section (#coded-
residual)) MUST be less than or equal to 8.
* The channel ordering MUST be equal to one defined in the section
channels bits (#channels-bits), i.e. the FLAC file MUST NOT need a
WAVEFORMATEXTENSIBLE_CHANNEL_MASK tag to describe the channel
ordering. See section channel mask (#channel-mask) 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
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 a 0
when other metadata blocks follow, otherwise it is a 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 invalid.
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.
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+=========+====================================================+
| Value | Metadata block type |
+=========+====================================================+
| 0 | Streaminfo |
+---------+----------------------------------------------------+
| 1 | Padding |
+---------+----------------------------------------------------+
| 2 | Application |
+---------+----------------------------------------------------+
| 3 | Seektable |
+---------+----------------------------------------------------+
| 4 | Vorbis comment |
+---------+----------------------------------------------------+
| 5 | Cuesheet |
+---------+----------------------------------------------------+
| 6 | Picture |
+---------+----------------------------------------------------+
| 7 - 126 | reserved |
+---------+----------------------------------------------------+
| 127 | invalid, to avoid confusion with a frame sync code |
+---------+----------------------------------------------------+
Table 2
8.2. Streaminfo
The streaminfo metadata block has information about the whole stream,
like sample rate, number of channels, total number of samples, etc.
It MUST be present as the first metadata block in the stream. Other
metadata blocks MAY follow. 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. 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 invalid. A decoder accepting
information from the streaminfo 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 the section on security considerations
(#security-considerations).
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+========+===================================================+
| 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 samples in stream. 'Samples' means inter- |
| | channel sample, i.e. one second of 44.1 kHz audio |
| | will have 44100 samples regardless of the number |
| | of channels. A value of zero here means the |
| | number of total samples is unknown. |
+--------+---------------------------------------------------+
| u(128) | MD5 signature of the unencoded audio data. This |
| | allows the decoder to determine if an error |
| | exists in the audio data even when the error does |
| | not result in an invalid bitstream. 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
lesser than the maximum block size. The last frame MUST have a block
size equal to or lesser than the maximum block size, it does not have
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to comply to the minimum block size because the block size of that
frame must be able to accommodate for 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 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 block, and the
actual smallest block size excluding the last block MAY be larger
than the minimum block size listed in the streaminfo 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 these 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 Vorbis comment metadata block (#vorbis-
comment) or an application metadata block (see application metadata
block (#application). This document does not define such metadata.
The MD5 signature is made by performing an MD5 transformation on 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 first 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 performing the MD5 transformation,
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.
So, 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 extention bits of 1st sample of 1st channel
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As another example, in the case of a 1-channel with 12-bit samples,
bits are lined-up as follows, showing the little-endian byte order
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 right place in 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) |
+------+---------------------------------------------+
Table 4
8.4. Application
The application metadata block is for use by third-party
applications. The only mandatory field is a 32-bit identifier. An
ID registry is being maintained at https://xiph.org/flac/id.html
(https://xiph.org/flac/id.html).
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+=======+===========================================+
| Data | Description |
+=======+===========================================+
| u(32) | Registered application ID. (Visit the |
| | registration page (https://xiph.org/flac/ |
| | id.html) to register an ID with FLAC.) |
+-------+-------------------------------------------+
| u(n) | Application data (n MUST be a multiple of |
| | 8, i.e. a whole number of bytes) |
+-------+-------------------------------------------+
Table 5
8.5. Seektable
The seektable 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
seektable 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 kilobyte of data. The number of
seek points is implied by the metadata header 'length' field, i.e.
equal to length / 18. There is also a special 'placeholder'
seekpoint which will be ignored by decoders but which can be used to
reserve space for future seek point insertion.
+============+==========================+
| Data | Description |
+============+==========================+
| SEEKPOINT+ | One or more seek points. |
+------------+--------------------------+
Table 6
A seektable is generally not usable for seeking in a FLAC file
embedded in a container, as such containers usually interleave FLAC
data with other data and the offsets used in seekpoints are those of
an unmuxed FLAC stream. Also, containers often provide their own
seeking methods. It is however possible to store the seektable in
the container along with other metadata when muxing a FLAC file, so
this stored seektable can be restored on demuxing the FLAC stream
into a standalone FLAC file.
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8.5.1. Seekpoint
+=======+==========================================================+
| Data | Description |
+=======+==========================================================+
| u(64) | Sample number of 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 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. 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 UTF-8 coded, and is not terminated in any way.
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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 field stored with a 4 byte
length. First, the 4 byte field length in bytes is stored as an
unsigned number, coded little-endian. The field itself is, like the
vendor string, UTF-8 coded, not terminated in any way.
Each field consists of a field name and a field content, 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) respectively.
The field contents can contain any UTF-8 character.
Note that the Vorbis comment as used in Vorbis allows for on the
order of 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
Except the one defined in the section channel mask (#channel-mask),
no standard field names are defined. In general, most FLAC playback
devices and software recognize 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 is it often the performer
* Album: name of the collection the current work belongs to
For a more comprehensive list of possible field names, the list of
tags used in the MusicBrainz project (http://picard-
docs.musicbrainz.org/en/variables/variables.html) is recommended.
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8.6.2. Channel mask
Besides fields containing information about the work itself, one
field is defined for technical reasons, of which the field name is
WAVEFORMATEXTENSIBLE_CHANNEL_MASK. This field is used to communicate
that the channels in a file differ from the default channels defined
in the section channels bits (#channels-bits). 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, stored in a hexadecimal representation preceded by 0x. The
flags only signal which channels are present, not in which order, so
if a file has to be encoded in which channels are ordered
differently, they have to be reordered. Please note that a file in
which the channel order is defined through the
WAVEFORMATEXTENSIBLE_CHANNEL_MASK is not streamable (see section
streamable subset (#streamable-subset)), as the field is not found in
each frame header. The mask bits can be found in the following table
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+============+=============================+
| 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 3 examples:
* if a file has a single channel, being a LFE channel, the Vorbis
comment field is WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x8
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* if a file has 4 channels, being front left, front right, top front
left and top front right, the Vorbis comment field is
WAVEFORMATEXTENSIBLE_CHANNEL_MASK=0x5003
* if an input has 4 channels, being back center, top front center,
front center and top rear center in that order, they 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 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 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
To either store the track and index point structure of a CD-DA along
with its audio or to provide a mechanism to store locations of
interest within a FLAC file, a cuesheet metadata block can be used.
Certain aspects of this metadata block follow directly from the CD-DA
specification, called Red Book, which is standardized as
[IEC.60908.1999]. For more information on the function and history
of these aspects, please refer to [IEC.60908.1999].
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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 Compact |
| | Disc, 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 in the |
| tracks | section cuesheet track (#cuesheet-track) 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 NUL characters. For CD-DA, this is a thirteen digit
number, followed by 115 NUL 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 silence and CD grabbing
software does not usually store it; additionally, the lead-in MUST be
at least two seconds but MAY be longer. For these reasons the lead-
in length is stored here so that the absolute position of the first
track can be computed. Note that the lead-in stored here is the
number of samples up to the first index point of the first track, not
necessarily to INDEX 01 of the first track; even the first track MAY
have INDEX 00 data.
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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 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 | number of structures as specified in the |
| index | section cuesheet track index point |
| points | (#cuesheet-track-index-point) 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 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).
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A track number of 0 is not allowed, because the CD-DA spec 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 NUL
characters MAY be used to denote 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 SHOULD 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, an 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 APIC frame in the ID3v2 specification. However,
contrary to the APIC frame in ID3v2, the media type and description
are prepended with a 4-byte length field instead of being null
delimited strings. A FLAC file MAY contain one or more picture
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metadata blocks.
Note that while the length fields for media type, description and
picture data are 4 bytes in length and could in theory code for a
size up to 4 GiB, the total metadata block size cannot exceed what
can be described by the metadata block header, i.e. 16 MiB.
The structure of a picture metadata block is enumerated in the
following table.
+========+==================================================+
| Data | Description |
+========+==================================================+
| u(32) | The picture type according to next table |
+--------+--------------------------------------------------+
| u(32) | The length of the media type string in bytes. |
+--------+--------------------------------------------------+
| u(n*8) | The media type string, in printable ASCII |
| | characters 0x20-0x7E. The media type MAY also |
| | be --> to signify that the data part is a URI of |
| | the picture instead of the picture data itself. |
+--------+--------------------------------------------------+
| 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, or 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 MAY use them
to decide to process a block or not (e.g. when selecting between
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different pictures 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 defined picture types. Values other
than those listed in the table are reserved. There MAY only be one
each of picture type 1 and 2 in a file. In general practice, many
FLAC playback devices and software display the contents of a picture
metadata block with picture type 3 (front cover) during playback, if
present.
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+=======+=================================================+
| Value | Picture type |
+=======+=================================================+
| 0 | Other |
+-------+-------------------------------------------------+
| 1 | PNG file icon of 32x32 pixels |
+-------+-------------------------------------------------+
| 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
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If not a picture but a URI is contained in this block, the following
points apply:
* The URI can be either in absolute or in relative form. If an 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 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, in
particular for security or privacy reasons.
9. Frame structure
Directly after the last metadata block, one or more frames follow.
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 the section on security
considerations (#security-considerations).
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 and it
is therefore 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 the 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 container mappings
(#container-mappings) for details.
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9.1. Frame header
Each frame MUST start on a byte boundary and starts 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
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 block size 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 uncommon
block size (#uncommon-block-size).
+=================+===========================================+
| 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
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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 or after the coded number if no uncommon block size was used.
See section uncommon sample rate (#uncommon-sample-rate).
+========+=======================================================+
| Value | Sample rate |
+========+=======================================================+
| 0b0000 | sample rate only stored in 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 | invalid |
+--------+-------------------------------------------------------+
Table 15
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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 lay-out 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 the section channel mask (#channel-mask) for details.
Note that even when such a different channel lay-out is specified
with a WAVEFORMATEXTENSIBLE_CHANNEL_MASK and the channel ordering in
the following table is overriden, the channels bits still contain the
actual number of channel coded in the frame. For details on the way
left/side, right/side and mid/side stereo are coded, see the section
on interchannel decorrelation (#interchannel-decorrelation).
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+==========+====================================================+
| 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 right/side |
| | 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.
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+=======+====================================================+
| Value | Bit depth |
+=======+====================================================+
| 0b000 | bit depth only stored in 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
The next bit is reserved and MUST be zero.
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.
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,
7 byte encoded.
When a frame number is encoded, the value MUST NOT be larger than
what fits a value 31 bits unencoded or 6 byte 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 of section 3 of [RFC3629], but instead
of using a character number, use a frame or sample number, and
instead of the table in section 3 of [RFC3629], use the extended
table below.
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+============================+=====================================+
| 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, a frame that belongs to a variable block size stream and
has exactly 51 billion samples preceding it, has its coded number
constructed as follows.
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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 non-consecutive
or otherwise invalid. See also the section on security
considerations (#security-considerations).
9.1.6. Uncommon block size
If the block size bits defined earlier in this section were 0b0110 or
0b0111 (uncommon block size minus 1 stored), this follows the coded
number as either an 8-bit or a 16-bit unsigned number coded big-
endian. A value of 65535 (corresponding to a block size of 65536) is
invalid 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 the section on the streaminfo metadata
block (#streaminfo).
9.1.7. Uncommon sample rate
Following the uncommon block size (or the coded number if no uncommon
block size is stored) is the sample rate, if the sample rate bits
were 0b1100, 0b1101 or 0b1110 (uncommon sample rate 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 streaminfo (#streaminfo) for details.
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9.1.8. Frame header CRC
Finally, after either the frame/sample number, an uncommon block size
or an uncommon sample rate, depending on whether the latter two are
stored, is an 8-bit CRC. 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 as subframes contain a bitstream
that does not necessarily has to be a whole number of bytes, only the
first subframe always 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 describing 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 has any wasted bits. If it is 0, the subframe doesn't have
any wasted bits and the subframe header is complete. If it is 1, the
subframe does have wasted bits and the number of wasted bits follows
unary coded.
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9.2.2. Wasted bits per sample
Certain file formats, like AIFF, can store audio samples with a bit
depth that is not an integer number of bytes 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 a sense that they contain no information, but are stored anyway.
The wasted bits-per-sample flag in a subframe header is set to 1 if
such wasted bits are present in that subframe. If this is the case,
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.
If 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 constant
subframe (#constant-subframe) 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, right/side or mid/side stereo, a decoder MUST
perform padding on the subframes before decorrelating 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 constant subframe (#constant-
subframe)) is larger than zero.
Besides audio files that have a certain number of wasted bits for the
whole file, there exist audio files 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, as to
slightly improve the compression of their otherwise lossless Meridian
Lossless Packing codec. There are also audio processors like
lossyWAV that enable users to improve compression of their files by a
lossless audio codec 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 a
subframe (e.g. the first part has 2 wasted bits and the second part
has 4 wasted bits) the subframe uses the lowest common denominator,
as otherwise it bits would be discarded and the process would not be
lossless.
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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 (#bit-depth-
bits), minus the number of wasted bits coded in the subframe header
(#wasted-bits-per-sample). If a subframe is a side subframe (see the
section on interchannel decorrelation (#interchannel-decorrelation)),
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 on Constant subframe (#constant-subframe) on how a sample
is stored unencoded. The number of samples that need to be stored in
a subframe is given 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. In the table is also a
derivation, which 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
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* 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 stage. See section on Constant subframe (#constant-
subframe) 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 | Encoded residual |
| residual | |
+----------+-------------------------------------------+
Table 21
As the 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 MUST be a sequential process within a subframe, as for
each sample, enough fully decoded previous samples are needed to
calculate the prediction.
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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 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 with 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 for each sample, enough decoded samples
are needed to calculate the prediction.
The table below defines how a linear predictor subframe appears in
the bitstream
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+==========+===============================================+
| 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 invalid). |
+----------+-----------------------------------------------+
| s(5) | Prediction right shift needed in bits. |
+----------+-----------------------------------------------+
| s(n) | Unencoded predictor coefficients (n = |
| | predictor coefficient precision * lpc order). |
+----------+-----------------------------------------------+
| Coded | Encoded residual |
| residual | |
+----------+-----------------------------------------------+
Table 22
See section on Constant subframe (#constant-subframe) on how the
warm-up samples are stored unencoded. The unencoded predictor
coefficients are stored the same way as the warm-up samples, but 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
section past changes (#restriction-of-lpc-shift-to-non-negative-
values) 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. 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
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+=============+=============================================+
| 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 amount 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 so that the block size is evenly
divisible by the number of partitions. This means for example that
for all odd block sizes, only partition order 0 is allowed. The
partition order also MUST be so 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, 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 at the start of
this section), 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.
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9.2.7.1. Escaped partition
If an escape code was used, the partition does not contain a
variable-length Rice coded residual, but 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 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 has 1 subtracted. This representation of signed
numbers is also known as zigzag encoding and the zigzag encoded
residual is called the folded residual.
Each folded residual sample is then split in 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 unary
followed by the least significant part binary.
For example, take a partition with Rice parameter 3 containing a
folded residual sample with 38 as value, which is 0b100110 in binary.
The most significant part is 0b100 (4) and is stored unary as
0b00001, the least significant part is 0b110 (6) and 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
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shifted right 1 bit (i.e. divided by two) to get the (unfolded)
residual value. An odd folded residual value is gets shifted right 1
bit and 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 examples (#examples) shows decoding of a complete coded
residual.
9.2.7.3. Residual sample value limit
All residual samples 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 then, 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 of which all
residual samples fall within said range, it MUST default to writing a
verbatim subframe. The appendix numerical considerations
(#numerical-considerations) explains in which circumstances residual
samples are already implicitly representable in said range and 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 int 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 int, and most library routines calculating an absolute
value have undefined behavior on 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 polynomial x^16 + x^15 + x^2 +
x^0. This CRC covers the whole frame excluding the 16-bit CRC,
including the sync code.
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10. Container mappings
The FLAC format can be used without any container as the FLAC format
already provides for a very thin transport layer. However, the
functionality of this transport is rather limited, and to be able to
combine FLAC audio with video, it needs to be encapsulated by a more
capable container. This presents a problem: the transport layer
provided by the FLAC format mixes data that belongs to the encoded
data (like block size and sample rate) with data that belongs to the
transport (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
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+=========+=========================================================+
| 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 block |
| bytes | |
+---------+---------------------------------------------------------+
| 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 an
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 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.
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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 by the last completed packet in
the frame. The sample numbering considers inter-channel samples. If
a page contains no packet end (e.g. when page only contains the start
of a large packet, which 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 which
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
[I-D.ietf-cellar-matroska]. 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 Matroska context.
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 a by finishing the current Matroska segment
and starting a new one with the new properties.
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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
https://github.com/xiph/flac/blob/master/doc/isoflac.txt
(https://github.com/xiph/flac/blob/master/doc/isoflac.txt) The
definitions document is summarized here.
The sample entry code is 'fLaC'. The channelcount and samplesize
fields in the sample entry follow from the values as found in the
FLAC stream. The samplerate field can be different, because FLAC can
carry audio with much higher sample rates than can be coded for in
the sample entry. When possible, the samplerate field should contain
the sample rate as found in the FLAC stream, shifted left by 16 bits
to get the 16.16 fixed point representation of the samplerate field.
When the FLAC stream contains a sample rate higher than can be coded,
the samplerate field contains the greatest expressible regular
division of the sample rate, e.g. 48000 for sample rates of 96kHz and
192kHz or 44100 for a sample rate of 88200Hz. When the FLAC stream
contain audio with an unusual sample rate that has no regular
division, the maximum value of 65535.0 Hz is used. As FLAC streams
with a high sample rate are common, a parser or decoder MUST read the
value from the FLAC streaminfo metadata block or a frame header to
determine the actual sample rate. The sample entry contains one
'FLAC specific box' with code 'dfLa'.
The FLAC specific box extends FullBox, with version number 0 and all
flags set to 0, and contains all FLAC data before the first audio
frame but fLaC ASCII signature (i.e. all metadata blocks).
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 MUST be dealt in a MP4 generic manner e.g. with several stsd
atoms and different sample-description-index values in the stsc atom.
Each FLAC frame is a single sample in the context of MP4 files.
11. Implementation status
This section records the status of known implementations of the FLAC
format, and is based on a proposal described in [RFC7942]. Please
note that the listing of any individual implementation here does not
imply endorsement by the IETF. Furthermore, no effort has been spent
to verify the information presented here that was supplied by IETF
contributors. This is not intended as, and must not be construed to
be, a catalog of available implementations or their features.
Readers are advised to note that other implementations may exist.
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A reference encoder and decoder implementation of the FLAC format
exists, known as libFLAC, maintained by Xiph.Org. It can be found at
https://xiph.org/flac/ (https://xiph.org/flac/) Note that while all
libFLAC components are licensed under 3-clause BSD, the flac and
metaflac command line tools often supplied together with libFLAC are
licensed under GPL.
Another completely independent implementation of both encoder and
decoder of the FLAC format is available in libavcodec, maintained by
FFmpeg, licensed under LGPL 2.1 or later. It can be found at
https://ffmpeg.org/ (https://ffmpeg.org/)
A list of other implementations and an overview of which parts of the
format they implement can be found here: https://github.com/ietf-wg-
cellar/flac-specification/wiki/Implementations (https://github.com/
ietf-wg-cellar/flac-specification/wiki/Implementations)
12. 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, as outlined in [RFC4732]. 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 to 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 in encoders are
typically rarer. Malformed audio streams MUST NOT cause the encoder
to misbehave because this would allow an attacker to attack
transcoding gateways.
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As with all compression algorithms, both encoding and decoding can
produce an output much larger than the input. In case of 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 byte in size, but codes for more than 2 megabyte 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 gigabyte 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 handling of extreme cases or impose reasonable limits
beyond the limits of this specification document. 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 fields
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, memory
for which must be allocated, parsers SHOULD 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. Applications MUST obtain explicit
user approval to retrieve resources via remote protocols and to
retrieve local resources not located in the same directory as the
FLAC file being processed.
Seeking in a FLAC stream that is not in a container relies on the
coded number in frame headers and optionally a seektable metadata
block. Parsers MUST employ thorough sanity checks on whether a found
coded number or seekpoint is at all possible. Without these checks,
seeking might get stuck in an infinite loop when numbers in frames
are non-consecutive or otherwise invalid, which could be used in
denial of service attacks.
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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 the FLAC decoder testbench (https://github.com/ietf-wg-cellar/
flac-test-files) for a non-exhaustive list of FLAC files with extreme
configurations which 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 for testing conformance
with this specification.
13. IANA Considerations
In accordance with the procedures set forth in [RFC6838], this
document registers one new media type, "audio/flac", as defined in
the following section.
13.1. Media type registration
The following information serves as the registration form for the
"audio/flac" media type. This media type is applicable for FLAC
audio packaged in its native container. FLAC audio packaged in
another container will take on the media type of its container, for
example audio/ogg when packaged in an Ogg container or video/mp4 when
packaged in a MP4 container alongside a video track.
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Type name: audio
Subtype name: flac
Required parameters: none
Optional parameters: none
Encoding considerations: as per THISRFC
Security considerations: see the security considerations in section 12 of THISRFC
Interoperability considerations: see the descriptions of past format changes in Appendix B of THISRFC
Published specification: THISRFC
Applications that use this media type: ffmpeg, apache, firefox
Fragment identifier considerations: none
Additional information:
Deprecated alias names for this type: audio/x-flac
Magic number(s): fLaC
File extension(s): flac
Macintosh file type code(s): none
Person & email address to contact for further information: IETF CELLAR WG
Intended usage: COMMON
Restrictions on usage: N/A
Author: IETF CELLAR WG
Change controller: Internet Engineering Task Force (mailto:iesg@ietf.org)
Provisional registration? (standards tree only): NO
14. Acknowledgments
FLAC owes much to the many people who have advanced the audio
compression field so freely. For instance:
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* A. J. Robinson for his work on Shorten; 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.
* S. W. Golomb and Robert F. Rice; their universal codes are used
by FLAC's entropy coder, see [Rice].
* N. Levinson and J. Durbin; the FLAC reference encoder (see
implementation status (#implementation-status)) uses an algorithm
developed and refined by them for determining the LPC coefficients
from the autocorrelation coefficients, see [Durbin].
* And of course, Claude Shannon, see [Shannon].
The FLAC format, the FLAC reference implementation and this document
were originally developed by Josh Coalson. While many others have
contributed since, this original effort is deeply appreciated.
15. Normative References
[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>.
[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>.
[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>.
16. Informative References
[Durbin] Durbin, J., "The Fitting of Time-Series Models",
DOI 10.2307/1401322, December 1959,
<https://www.jstor.org/stable/1401322>.
[FIR] "Finite impulse response - Wikipedia",
<https://en.wikipedia.org/wiki/Finite_impulse_response>.
[HPL-1999-144]
Hans, M. and RW. Schafer, "Lossless Compression of Digital
Audio", DOI 10.1109/79.939834, November 1999,
<https://www.hpl.hp.com/techreports/1999/HPL-
1999-144.pdf>.
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[I-D.ietf-cellar-matroska]
Lhomme, S., Bunkus, M., and D. Rice, "Matroska Media
Container Format Specifications", Work in Progress,
Internet-Draft, draft-ietf-cellar-matroska-16, 1 April
2023, <https://datatracker.ietf.org/doc/html/draft-ietf-
cellar-matroska-16>.
[IEC.60908.1999]
International Electrotechnical Commission, "Audio
recording - Compact disc digital audio system",
IEC International standard 60908 second edition, 1999.
[ISRC-handbook]
International ISRC Registration Authority, "International
Standard Recording Code (ISRC) Handbook, 4th edition",
2021, <https://www.ifpi.org/isrc_handbook/>.
[LinearPrediction]
"Linear prediction - Wikipedia",
<https://en.wikipedia.org/wiki/Linear_prediction>.
[RFC3533] Pfeiffer, S., "The Ogg Encapsulation Format Version 0",
RFC 3533, DOI 10.17487/RFC3533, May 2003,
<https://www.rfc-editor.org/info/rfc3533>.
[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>.
[RFC6838] Freed, N., Klensin, J., and T. Hansen, "Media Type
Specifications and Registration Procedures", BCP 13,
RFC 6838, DOI 10.17487/RFC6838, January 2013,
<https://www.rfc-editor.org/info/rfc6838>.
[RFC7942] Sheffer, Y. and A. Farrel, "Improving Awareness of Running
Code: The Implementation Status Section", BCP 205,
RFC 7942, DOI 10.17487/RFC7942, July 2016,
<https://www.rfc-editor.org/info/rfc7942>.
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[Rice] Rice, RF. and JR. Plaunt, "Adaptive Variable-Length Coding
for Efficient Compression of Spacecraft Television Data",
DOI 10.1109/TCOM.1971.1090789, December 1971,
<https://ieeexplore.ieee.org/document/1090789>.
[Shannon] Shannon, CE., "Communication in the Presence of Noise",
DOI 10.1109/JRPROC.1949.232969, January 1949,
<https://ieeexplore.ieee.org/document/1697831>.
[VarLengthCode]
"Variable-length code - Wikipedia",
<https://en.wikipedia.org/wiki/Variable-length_code>.
[Vorbis] Xiph.Org, "Ogg Vorbis I format specification: comment
field and header specification",
<https://xiph.org/vorbis/doc/v-comment.html>.
[robinson-tr156]
Robinson, T., "SHORTEN: Simple lossless and near-lossless
waveform compression", December 1994,
<https://mi.eng.cam.ac.uk/reports/abstracts/
robinson_tr156.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 large enough data types. 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 section provides guidelines for choosing the right
data type in each step of encoding and decoding FLAC files.
A.1. Determining 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.
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If for example 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, an
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.
In case of multiplication, the number of bits needed for the result
is the size of the first variable plus the size of the second
variable, but counting only one sign bit if working with signed data
types. If for example a 16-bit signed integer is multiplied by a
16-bit signed integer, the result needs at least 31 bits to store
without overflowing.
A.2. Stereo decorrelation
When stereo decorrelation is used, the side channel will have one
extra bit of bit depth, see section on Interchannel Decorrelation
(#interchannel-decorrelation).
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
bit, the decoding of a side subframe in such a file will need a data
type with at least 17 bit 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 bit.
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 maximal possible value for these
is calculated as described in the section determining necessary data
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type size (#determining-necessary-data-type-size) 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 25
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, calculation is a little more
complicated. Each prediction is a 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. As both are signed numbers and only one 'sign bit' is
necessary, 1 bit can be subtracted. 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 the section on interchannel
decorrelation (#interchannel-decorrelation)), 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 - 1) + ceil(log2(12)) = 43 bits. As another example, with a
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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 - 1) +
ceil(log2(8)) = 31 bits.
A.4. Residual
As stated in the section coded residual (#coded-residual), an encoder
must make sure residual samples are representable by a 32-bit
integer, signed two's complement, excluding the most negative value.
Continuing 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 int, as that satisfies both mentioned
criteria.
For the residual of a fixed predictor, the maximum size of a residual
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, 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 31 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 31 - 10
+ 1 = 22 bits, which means it is not necessary to check whether the
residuals fit a 32-bit signed integer.
As another example, when encoding 32-bit PCM with fixed predictors,
all predictor orders must be checked. While the 0-order fixed
predictor is guaranteed to have residuals that fit a 32-bit signed
int, it might produce a residual being the most negative
representable value of that 32-bit signed int.
Note that on decoding, while the residual samples 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 wide
enough signed integer data type like on encoding.
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A.5. Rice coding
When folding (i.e. zig-zag 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. Many implementations
however choose to require one extra bit of data type size so zig-zag
encoding can happen in one step and without a cast instead of the
procedure described in the previous sentence.
Appendix B. Past format changes
This informational appendix documents what changes were 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 (the
reference encoder/decoder) 1.0 in July 2001. Only changes made since
this first stable release are considered in this appendix. Changes
made to the FLAC streamble subset definition (see section streamable
subset (#streamable-subset)) are not considered.
B.1. Addition of block size strategy flag
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, either by comparing the minimum and
maximum block size in the STREAMINFO metadata block (which are equal
in case of a fixed block size stream), or, if a decoder did not
receive a STREAMINFO metadata block, by detecting a change of block
size during a stream, which could in theory not happen at all. As
the meaning of the coded number in the frame header depends on
whether or not a stream is 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 block size strategy flag. See also the
section frame header (#frame-header).
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Along with the addition of a new flag, the meaning of the block size
bits (#block-size-bits) was subtly changed. Initially, block size
bits 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 the change
these restrictions were lifted and 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 coded residual (#coded-residual) 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 existing
implementation would be rendered non-compliant by this change.
B.3. Addition of 5-bit Rice parameter
One significant addition to the format was the residual coding method
using a 5-bit Rice parameter. Prior to publication of this addition
in July 2007, there was only one residual coding method specified, a
partitioned Rice code with a 4-bit Rice parameter. The range offered
by this proved too small when encoding 24-bit PCM, therefore a second
residual coding method was specified identical to the first but with
a 5-bit Rice parameter.
B.4. Restriction of LPC shift to non-negative values
As stated in section linear predictor subframe (#linear-predictor-
subframe), the predictor right shift is a number signed two's
complement, which MUST NOT be negative. This is because right
shifting a number 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 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
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at that point were not able to handle such subframes.
Appendix C. Interoperability considerations
As documented in appendix past format changes (#past-format-changes),
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 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 finished,
consult this web page (https://github.com/ietf-wg-cellar/flac-
specification/wiki/Interoperability-considerations) for more up-to-
date information.
C.1. Features outside of streamable subset
As described in section streamable subset (#streamable-subset), 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 is explained in section
addition of block size strategy flag (#addition-of-block-size-
strategy-flag), there have been some changes to the way variable
block size streams were encoded. Because of this, maximum
compatibility with decoders is achieved when FLAC files are created
using fixed block size streams.
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C.3. 5-bit Rice parameter
As the addition of the 5-bit Rice parameter as described in section
addition of 5-bit Rice parameter (#addition-of-5-bit-rice-parameter)
was 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 a 5-bit Rice parameters.
Therefore, maximum compatibility with decoders is achieved when FLAC
files containing audio with a bit depth of 16 bits or lower are
created without any use of 5-bit Rice parameters.
C.4. Rice escape code
Escapes Rice partitions are only seldom used as it turned out their
use provides only very small compression improvement. As many
encoders therefore do not use these by default or are not capable of
producing them at all, it is likely many decoder implementation 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 except for the last block. Therefore, maximum
compatibility with decoders is achieved when creating FLAC files
using common block sizes as listed in section block size bits
(#block-size-bits) for all but the last block.
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 bit. There is however 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 bit, which is companded to 10 bit
* 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
FLAC can store 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
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each sample with zero bits to a bit depth that is a whole byte. FLAC
will detect these wasted bits. This transformation leaves no
ambiguity in how it can be reversed and is thus lossless. See
section wasted bits per sample (#wasted-bits-per-sample) 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.
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 bit. Many implementations support bit
depths up to 24 bit but no higher. Consult this web page
(https://github.com/ietf-wg-cellar/flac-specification/wiki/
Interoperability-considerations) 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 that cannot be satisfied
without sacrificing the lossless nature of the FLAC format.
From a non-exhaustive inquiry, it seems that a non-negligible amount
of players, among those especially hardware players, does not support
audio with 3 or more channels or sample rates other than those
considered common, see section sample rate bits (#sample-rate-bits).
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 channel mask
(#channel-mask)). This too is a interoperability consideration that
cannot be satisfied without sacrificing the lossless nature of the
FLAC format.
Appendix D. Examples
This informational appendix contains short example FLAC files which
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 lay-out of several metadata blocks and
subframe types and the implications of certain aspects (for example
wasted bits and stereo decorrelation) on this lay-out.
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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 https://github.com/ietf-wg-cellar/
flac-specification (https://github.com/ietf-wg-cellar/flac-
specification).
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 1 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
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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 byte | 0x664C6143 | fLaC |
+--------+--------+------------+---------------------------+
| 0x04+0 | 1 bit | 0b1 | Last metadata block |
+--------+--------+------------+---------------------------+
| 0x04+1 | 7 bit | 0b0000000 | Streaminfo metadata block |
+--------+--------+------------+---------------------------+
| 0x05+0 | 3 byte | 0x000022 | Length 34 byte |
+--------+--------+------------+---------------------------+
Table 26
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. As can be seen 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
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+========+=========+====================+=========================+
| Start | Length | Contents | Description |
+========+=========+====================+=========================+
| 0x08+0 | 2 byte | 0x1000 | Min. block size 4096 |
+--------+---------+--------------------+-------------------------+
| 0x0a+0 | 2 byte | 0x1000 | Max. block size 4096 |
+--------+---------+--------------------+-------------------------+
| 0x0c+0 | 3 byte | 0x00000f | Min. frame size 15 byte |
+--------+---------+--------------------+-------------------------+
| 0x0f+0 | 3 byte | 0x00000f | Max. frame size 15 byte |
+--------+---------+--------------------+-------------------------+
| 0x12+0 | 20 bit | 0x0ac4, 0b0100 | Sample rate 44100 hertz |
+--------+---------+--------------------+-------------------------+
| 0x14+4 | 3 bit | 0b001 | 2 channels |
+--------+---------+--------------------+-------------------------+
| 0x14+7 | 5 bit | 0b01111 | Sample bit depth 16 |
+--------+---------+--------------------+-------------------------+
| 0x15+4 | 36 bit | 0b0000, 0x00000001 | Total no. of samples 1 |
+--------+---------+--------------------+-------------------------+
| 0x1a | 16 byte | (...) | MD5 signature |
+--------+---------+--------------------+-------------------------+
Table 27
The minimum and maximum block size 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 sum (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.
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+========+========+=================+===============================+
| Start | Length | Contents | Description |
+========+========+=================+===============================+
| 0x2a+0 | 15 bit | 0xff, 0b1111100 | frame sync |
+--------+--------+-----------------+-------------------------------+
| 0x2b+7 | 1 bit | 0b0 | block size strategy |
+--------+--------+-----------------+-------------------------------+
| 0x2c+0 | 4 bit | 0b0110 | 8-bit block size |
| | | | further down |
+--------+--------+-----------------+-------------------------------+
| 0x2c+4 | 4 bit | 0b1001 | sample rate 44.1kHz |
+--------+--------+-----------------+-------------------------------+
| 0x2d+0 | 4 bit | 0b0001 | stereo, no |
| | | | decorrelation |
+--------+--------+-----------------+-------------------------------+
| 0x2d+4 | 3 bit | 0b100 | bit depth 16 bit |
+--------+--------+-----------------+-------------------------------+
| 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 28
As the stream is a fixed block size stream, the number at 0x2e
contains a frame number. As the value is smaller than 128, only 1
byte is used for the encoding.
At byte 0x31 the subframe header of the first subframe starts, it is
broken down in the following table.
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+========+========+================+=========================+
| Start | Length | Contents | Description |
+========+========+================+=========================+
| 0x31+0 | 1 bit | 0b0 | mandatory 0 bit |
+--------+--------+----------------+-------------------------+
| 0x31+1 | 6 bit | 0b000001 | verbatim subframe |
+--------+--------+----------------+-------------------------+
| 0x31+7 | 1 bit | 0b1 | wasted bits present |
+--------+--------+----------------+-------------------------+
| 0x32+0 | 2 bit | 0b01 | 2 wasted bits |
+--------+--------+----------------+-------------------------+
| 0x32+2 | 14 bit | 0b011000, 0xfd | 14-bit unencoded sample |
+--------+--------+----------------+-------------------------+
Table 29
As the wasted bits flag is 1 in this subframe, an unary coded number
follows. Starting at 0x32, we see 0b01, which unary codes for 1,
meaning we have 2 wasted bits in this subframe.
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 bit, but as the subframe
header signals 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 bit (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, it is broken down in the
following table.
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+========+========+==============+=========================+
| Start | Length | Contents | Description |
+========+========+==============+=========================+
| 0x34+0 | 1 bit | 0b0 | mandatory 0 bit |
+--------+--------+--------------+-------------------------+
| 0x34+1 | 6 bit | 0b000001 | verbatim subframe |
+--------+--------+--------------+-------------------------+
| 0x34+7 | 1 bit | 0b1 | wasted bits present |
+--------+--------+--------------+-------------------------+
| 0x35+0 | 4 bit | 0b0001 | 4 wasted bits |
+--------+--------+--------------+-------------------------+
| 0x35+4 | 12 bit | 0b0010, 0x8b | 12-bit unencoded sample |
+--------+--------+--------------+-------------------------+
Table 30
Here the wasted bits flag is also one, but the unary coded number
that follows it is 4 bit long, indicating 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 do 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, we line up the samples interleaved, byte-
aligned, little endian, signed two's complement. The first sample,
the value of which was 25588 translates to 0xf463, the second sample
had a value of 10416 which translates to 0xb028. When MD5 summing
0xf463b028, we get the MD5 sum 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 seektable, Vorbis comment and padding metadata
block.
D.2.1. Example file 2 in hexadecimal representation
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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
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D.2.3. Streaminfo metadata block
Most of the streaminfo 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 byte | 0x0010 | Min. block size 16 |
+--------+--------+------------+-----------------------------+
| 0x0a+0 | 2 byte | 0x0010 | Max. block size 16 |
+--------+--------+------------+-----------------------------+
| 0x0c+0 | 3 byte | 0x000017 | Min. frame size 23 byte |
+--------+--------+------------+-----------------------------+
| 0x0f+0 | 3 byte | 0x000044 | Max. frame size 68 byte |
+--------+--------+------------+-----------------------------+
| 0x15+4 | 36 bit | 0b0000, | Total no. of samples 19 |
| | | 0x00000013 | |
+--------+--------+------------+-----------------------------+
| 0x1a | 16 | (...) | MD5 signature |
| | byte | | |
+--------+--------+------------+-----------------------------+
Table 31
This time, the minimum and maximum block sizes are reflected in the
file: there is one block of 16 samples, the last block (which has 3
samples) is not considered for the minimum block size. The MD5
signature is 0xd5b0 5649 75e9 8b8d 8b93 0422 757b 8103, this will be
verified at the end of this example.
D.2.4. Seektable
The seektable 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 seektable metadata block is broken down in the
following table.
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+========+========+====================+================+
| Start | Length | Contents | Description |
+========+========+====================+================+
| 0x2a+0 | 1 bit | 0b0 | Not the last |
| | | | metadata block |
+--------+--------+--------------------+----------------+
| 0x2a+1 | 7 bit | 0b0000011 | Seektable |
| | | | metadata block |
+--------+--------+--------------------+----------------+
| 0x2b+0 | 3 byte | 0x000012 | Length 18 byte |
+--------+--------+--------------------+----------------+
| 0x2e+0 | 8 byte | 0x0000000000000000 | Seekpoint to |
| | | | sample 0 |
+--------+--------+--------------------+----------------+
| 0x36+0 | 8 byte | 0x0000000000000000 | Seekpoint to |
| | | | offset 0 |
+--------+--------+--------------------+----------------+
| 0x3e+0 | 2 byte | 0x0010 | Seekpoint to |
| | | | block size 16 |
+--------+--------+--------------------+----------------+
Table 32
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.
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+========+=========+============+===============================+
| Start | Length | Contents | Description |
+========+=========+============+===============================+
| 0x40+0 | 1 bit | 0b0 | Not the last metadata block |
+--------+---------+------------+-------------------------------+
| 0x40+1 | 7 bit | 0b0000100 | Vorbis comment metadata block |
+--------+---------+------------+-------------------------------+
| 0x41+0 | 3 byte | 0x00003a | Length 58 byte |
+--------+---------+------------+-------------------------------+
| 0x44+0 | 4 byte | 0x20000000 | Vendor string length 32 byte |
+--------+---------+------------+-------------------------------+
| 0x48+0 | 32 byte | (...) | Vendor string |
+--------+---------+------------+-------------------------------+
| 0x68+0 | 4 byte | 0x01000000 | Number of fields 1 |
+--------+---------+------------+-------------------------------+
| 0x6c+0 | 4 byte | 0x0e000000 | Field length 14 byte |
+--------+---------+------------+-------------------------------+
| 0x70+0 | 14 byte | (...) | Field contents |
+--------+---------+------------+-------------------------------+
Table 33
The vendor string is reference libFLAC 1.3.3 20190804, the field
contents of the only field is TITLE=שלום. 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 bit | 0b0000001 | Padding metadata block |
+--------+--------+----------------+------------------------+
| 0x7f+0 | 3 byte | 0x000006 | Length 6 byte |
+--------+--------+----------------+------------------------+
| 0x82+0 | 6 byte | 0x000000000000 | Padding bytes |
+--------+--------+----------------+------------------------+
Table 34
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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 bit | 0xff, 0b1111100 | frame sync |
+--------+--------+-----------------+-------------------------------+
| 0x89+7 | 1 bit | 0b0 | block size strategy |
+--------+--------+-----------------+-------------------------------+
| 0x8a+0 | 4 bit | 0b0110 | 8-bit block size |
| | | | further down |
+--------+--------+-----------------+-------------------------------+
| 0x8a+4 | 4 bit | 0b1001 | sample rate 44.1kHz |
+--------+--------+-----------------+-------------------------------+
| 0x8b+0 | 4 bit | 0b1001 | right-side stereo |
+--------+--------+-----------------+-------------------------------+
| 0x8b+4 | 3 bit | 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 35
The first subframe starts at byte 0x8f, 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 bit
to 17 bit. This is most clearly present in the unencoded warm-up
sample.
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+========+========+=============+===========================+
| Start | Length | Contents | Description |
+========+========+=============+===========================+
| 0x8f+0 | 1 bit | 0b0 | mandatory 0 bit |
+--------+--------+-------------+---------------------------+
| 0x8f+1 | 6 bit | 0b001001 | fixed subframe, 1st order |
+--------+--------+-------------+---------------------------+
| 0x8f+7 | 1 bit | 0b0 | no wasted bits present |
+--------+--------+-------------+---------------------------+
| 0x90+0 | 17 bit | 0x0867, 0b0 | unencoded warm-up sample |
+--------+--------+-------------+---------------------------+
Table 36
The coded residual is broken down in the following table. All
quotients are unary coded, all remainders are stored unencoded with a
number of bits specified by the Rice parameter.
+========+========+=================+=================+
| Start | Length | Contents | Description |
+========+========+=================+=================+
| 0x92+1 | 2 bit | 0b00 | Rice code with |
| | | | 4-bit parameter |
+--------+--------+-----------------+-----------------+
| 0x92+3 | 4 bit | 0b0000 | Partition order |
| | | | 0 |
+--------+--------+-----------------+-----------------+
| 0x92+7 | 4 bit | 0b1011 | Rice parameter |
| | | | 11 |
+--------+--------+-----------------+-----------------+
| 0x93+3 | 4 bit | 0b0001 | Quotient 3 |
+--------+--------+-----------------+-----------------+
| 0x93+7 | 11 bit | 0b00011110100 | Remainder 244 |
+--------+--------+-----------------+-----------------+
| 0x95+2 | 2 bit | 0b01 | Quotient 1 |
+--------+--------+-----------------+-----------------+
| 0x95+4 | 11 bit | 0b01000100001 | Remainder 545 |
+--------+--------+-----------------+-----------------+
| 0x96+7 | 2 bit | 0b01 | Quotient 1 |
+--------+--------+-----------------+-----------------+
| 0x97+1 | 11 bit | 0b00110011000 | Remainder 408 |
+--------+--------+-----------------+-----------------+
| 0x98+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x98+5 | 11 bit | 0b11101011101 | Remainder 1885 |
+--------+--------+-----------------+-----------------+
| 0x9a+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
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| 0x9a+1 | 11 bit | 0b11101110000 | Remainder 1904 |
+--------+--------+-----------------+-----------------+
| 0x9b+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x9b+5 | 11 bit | 0b10101101111 | Remainder 1391 |
+--------+--------+-----------------+-----------------+
| 0x9d+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x9d+1 | 11 bit | 0b11000000000 | Remainder 1536 |
+--------+--------+-----------------+-----------------+
| 0x9e+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0x9e+5 | 11 bit | 0b10000010111 | Remainder 1047 |
+--------+--------+-----------------+-----------------+
| 0xa0+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa0+1 | 11 bit | 0b10010101110 | Remainder 1198 |
+--------+--------+-----------------+-----------------+
| 0xa1+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa1+5 | 11 bit | 0b01100100001 | Remainder 801 |
+--------+--------+-----------------+-----------------+
| 0xa3+0 | 13 bit | 0b0000000000001 | Quotient 12 |
+--------+--------+-----------------+-----------------+
| 0xa4+5 | 11 bit | 0b11011100111 | Remainder 1767 |
+--------+--------+-----------------+-----------------+
| 0xa6+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa6+1 | 11 bit | 0b01001110111 | Remainder 631 |
+--------+--------+-----------------+-----------------+
| 0xa7+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa7+5 | 11 bit | 0b01000100100 | Remainder 548 |
+--------+--------+-----------------+-----------------+
| 0xa9+0 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xa9+1 | 11 bit | 0b01000010101 | Remainder 533 |
+--------+--------+-----------------+-----------------+
| 0xaa+4 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+-----------------+-----------------+
| 0xaa+5 | 11 bit | 0b00100001100 | Remainder 268 |
+--------+--------+-----------------+-----------------+
Table 37
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
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and remainder, and undoing the zig-zag encoding. For example, the
value of the first zig-zag encoded residual sample is 3 * 2^11 + 244
= 6388. As this is an even number, the zig-zag 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 | Zig-zag 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 38
It can be calculated that using a Rice code is in this case 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.
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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,
this means adding the residual value 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 39
With this, decoding of the first subframe is complete. Decoding of
the second subframe is very similar, as it also uses a fixed
predictor of order 1, so this is left as an exercise for the reader,
results are in the next table. The next step is stereo
decorrelation, which is done in the following table. As the stereo
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decorrelation is right-side, in which the actual ordering of the
subframes 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 40
As the second subframe ends byte-aligned, no padding bits follow it.
Finally, the last 2 bytes of the frame contain the frame CRC.
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D.2.8. Second audio frame
The second audio frame is very similar to the frame decoded in the
first example, but this time not 1 but 3 samples are present.
The frame header starts at position 0xcc and is broken down in the
following table.
+========+========+=================+===============================+
| Start | Length | Contents | Description |
+========+========+=================+===============================+
| 0xcc+0 | 15 bit | 0xff, 0b1111100 | frame sync |
+--------+--------+-----------------+-------------------------------+
| 0xcd+7 | 1 bit | 0b0 | block size strategy |
+--------+--------+-----------------+-------------------------------+
| 0xce+0 | 4 bit | 0b0110 | 8-bit block size |
| | | | further down |
+--------+--------+-----------------+-------------------------------+
| 0xce+4 | 4 bit | 0b1001 | sample rate 44.1kHz |
+--------+--------+-----------------+-------------------------------+
| 0xcf+0 | 4 bit | 0b0001 | stereo, no |
| | | | decorrelation |
+--------+--------+-----------------+-------------------------------+
| 0xcf+4 | 3 bit | 0b100 | bit depth 16 bit |
+--------+--------+-----------------+-------------------------------+
| 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 41
The first subframe starts at 0xd3+0 and is broken down in the
following table.
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+========+========+==========+=========================+
| Start | Length | Contents | Description |
+========+========+==========+=========================+
| 0xd3+0 | 1 bit | 0b0 | mandatory 0 bit |
+--------+--------+----------+-------------------------+
| 0xd3+1 | 6 bit | 0b000001 | verbatim subframe |
+--------+--------+----------+-------------------------+
| 0xd3+7 | 1 bit | 0b0 | no wasted bits present |
+--------+--------+----------+-------------------------+
| 0xd4+0 | 16 bit | 0xc382 | 16-bit unencoded sample |
+--------+--------+----------+-------------------------+
| 0xd6+0 | 16 bit | 0xc40b | 16-bit unencoded sample |
+--------+--------+----------+-------------------------+
| 0xd8+0 | 16 bit | 0xc14a | 16-bit unencoded sample |
+--------+--------+----------+-------------------------+
Table 42
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 bit | 0b000001 | verbatim subframe |
+--------+--------+-------------------+-------------------------+
| 0xda+7 | 1 bit | 0b1 | wasted bits present |
+--------+--------+-------------------+-------------------------+
| 0xdb+0 | 1 bit | 0b1 | 1 wasted bit |
+--------+--------+-------------------+-------------------------+
| 0xdb+1 | 15 bit | 0b110111001001000 | 15-bit unencoded sample |
+--------+--------+-------------------+-------------------------+
| 0xdd+0 | 15 bit | 0b110111010000001 | 15-bit unencoded sample |
+--------+--------+-------------------+-------------------------+
| 0xde+7 | 15 bit | 0b110110110011111 | 15-bit unencoded sample |
+--------+--------+-------------------+-------------------------+
Table 43
As this subframe has 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 follows at 0xe1+0.
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D.2.9. MD5 checksum verification
All samples in the file have been decoded, we can now verify the MD5
sum. 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 MD5sum 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
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+========+=========+====================+=========================+
| Start | Length | Contents | Description |
+========+=========+====================+=========================+
| 0x0c+0 | 3 byte | 0x00001f | Min. frame size 31 byte |
+--------+---------+--------------------+-------------------------+
| 0x0f+0 | 3 byte | 0x00001f | Max. frame size 31 byte |
+--------+---------+--------------------+-------------------------+
| 0x12+0 | 20 bit | 0x07d0, 0x0000 | Sample rate 32000 hertz |
+--------+---------+--------------------+-------------------------+
| 0x14+4 | 3 bit | 0b000 | 1 channel |
+--------+---------+--------------------+-------------------------+
| 0x14+7 | 5 bit | 0b00111 | Sample bit depth 8 bit |
+--------+---------+--------------------+-------------------------+
| 0x15+4 | 36 bit | 0b0000, 0x00000018 | Total no. of samples 24 |
+--------+---------+--------------------+-------------------------+
| 0x1a | 16 byte | (...) | MD5 signature |
+--------+---------+--------------------+-------------------------+
Table 44
D.3.4. Audio frame
The frame header starts at position 0x2a and is broken down in the
following table.
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+========+========+=================+===============================+
| Start | Length | Contents | Description |
+========+========+=================+===============================+
| 0x2a+0 | 15 bit | 0xff, 0b1111100 | Frame sync |
+--------+--------+-----------------+-------------------------------+
| 0x2b+7 | 1 bit | 0b0 | Block size strategy |
+--------+--------+-----------------+-------------------------------+
| 0x2c+0 | 4 bit | 0b0110 | 8-bit block size |
| | | | further down |
+--------+--------+-----------------+-------------------------------+
| 0x2c+4 | 4 bit | 0b1000 | Sample rate 32kHz |
+--------+--------+-----------------+-------------------------------+
| 0x2d+0 | 4 bit | 0b0000 | Mono audio (1 |
| | | | channel) |
+--------+--------+-----------------+-------------------------------+
| 0x2d+4 | 3 bit | 0b001 | Bit depth 8 bit |
+--------+--------+-----------------+-------------------------------+
| 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 45
The first and only subframe starts at byte 0x31, it is broken down in
the following table, without the coded residual.
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+========+========+==========+=====================+
| Start | Length | Contents | Description |
+========+========+==========+=====================+
| 0x31+0 | 1 bit | 0b0 | Mandatory 0 bit |
+--------+--------+----------+---------------------+
| 0x31+1 | 6 bit | 0b100010 | Linear prediction |
| | | | subframe, 3rd order |
+--------+--------+----------+---------------------+
| 0x31+7 | 1 bit | 0b0 | No wasted bits |
| | | | present |
+--------+--------+----------+---------------------+
| 0x32+0 | 8 bit | 0x00 | Unencoded warm-up |
| | | | sample 0 |
+--------+--------+----------+---------------------+
| 0x33+0 | 8 bit | 0x4f | Unencoded warm-up |
| | | | sample 79 |
+--------+--------+----------+---------------------+
| 0x34+0 | 8 bit | 0x6f | Unencoded warm-up |
| | | | sample 111 |
+--------+--------+----------+---------------------+
| 0x35+0 | 4 bit | 0b0011 | Coefficient |
| | | | precision 4 bit |
+--------+--------+----------+---------------------+
| 0x35+4 | 5 bit | 0b00010 | Prediction right |
| | | | shift 2 |
+--------+--------+----------+---------------------+
| 0x36+1 | 4 bit | 0b0111 | Predictor |
| | | | coefficient 7 |
+--------+--------+----------+---------------------+
| 0x36+5 | 4 bit | 0b1010 | Predictor |
| | | | coefficient -6 |
+--------+--------+----------+---------------------+
| 0x37+1 | 4 bit | 0b0010 | Predictor |
| | | | coefficient 2 |
+--------+--------+----------+---------------------+
Table 46
The data stream continues with the coded residual, which is broken
down in the following table. Residual partition 3 and 4 are left as
an exercise for the reader.
+========+========+==========+======================================+
| Start | Length | Contents | Description |
+========+========+==========+======================================+
| 0x37+5 | 2 bit | 0b00 | Rice-coded residual, |
| | | | 4-bit parameter |
+--------+--------+----------+--------------------------------------+
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| 0x37+7 | 4 bit | 0b0010 | Partition order 2 |
+--------+--------+----------+--------------------------------------+
| 0x38+3 | 4 bit | 0b0011 | Rice parameter 3 |
+--------+--------+----------+--------------------------------------+
| 0x38+7 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+----------+--------------------------------------+
| 0x39+0 | 3 bit | 0b110 | Remainder 6 |
+--------+--------+----------+--------------------------------------+
| 0x39+3 | 1 bit | 0b1 | Quotient 0 |
+--------+--------+----------+--------------------------------------+
| 0x39+4 | 3 bit | 0b001 | Remainder 1 |
+--------+--------+----------+--------------------------------------+
| 0x39+7 | 4 bit | 0b0001 | Quotient 3 |
+--------+--------+----------+--------------------------------------+
| 0x3a+3 | 3 bit | 0b001 | Remainder 1 |
+--------+--------+----------+--------------------------------------+
| 0x3a+6 | 4 bit | 0b1111 | No Rice parameter, |
| | | | escape code |
+--------+--------+----------+--------------------------------------+
| 0x3b+2 | 5 bit | 0b00101 | Partition encoded |
| | | | with 5 bits |
+--------+--------+----------+--------------------------------------+
| 0x3b+7 | 5 bit | 0b10110 | Residual -10 |
+--------+--------+----------+--------------------------------------+
| 0x3c+4 | 5 bit | 0b11010 | Residual -6 |
+--------+--------+----------+--------------------------------------+
| 0x3d+1 | 5 bit | 0b00010 | Residual 2 |
+--------+--------+----------+--------------------------------------+
| 0x3d+6 | 5 bit | 0b01000 | Residual 8 |
+--------+--------+----------+--------------------------------------+
| 0x3e+3 | 5 bit | 0b01000 | Residual 8 |
+--------+--------+----------+--------------------------------------+
| 0x3f+0 | 5 bit | 0b00110 | Residual 6 |
+--------+--------+----------+--------------------------------------+
| 0x3f+5 | 4 bit | 0b0010 | Rice parameter 2 |
+--------+--------+----------+--------------------------------------+
| 0x40+1 | 22 bit | (...) | Residual partition 3 |
+--------+--------+----------+--------------------------------------+
| 0x42+7 | 4 bit | 0b0001 | Rice parameter 1 |
+--------+--------+----------+--------------------------------------+
| 0x43+3 | 23 bit | (...) | Residual partition 4 |
+--------+--------+----------+--------------------------------------+
Table 47
The frame ends with 6 padding bits and a 2 byte frame CRC
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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 a subframe or decode a subframe with
parallel threads.
The following table breaks down the calculation of 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 bit: 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 |
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+-----------+---------------------+-----------+--------------+
| 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 48
Lining all these samples up, we get the following input for the MD5
summing process.
0x004F 6F4E 08C3 A6BC F32A 4335 0DE5 D2DA F40E 1813 06FC FB00
Which indeed results in the MD5 signature found in the streaminfo
metadata block.
Authors' Addresses
Martijn van Beurden
Netherlands
Email: mvanb1@gmail.com
Andrew Weaver
Email: theandrewjw@gmail.com
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