This is a detailed description of the FLAC format. There is also a companion document that describes
FLAC-to-Ogg mapping.
For a user-oriented overview, see About the FLAC Format.
Table of Contents
FLAC owes much to the many people who have advanced the audio compression field so freely. For instance:
It is a known fact that no algorithm can losslessly compress all possible input, so most compressors restrict themselves to a useful domain and try to work as well as possible within that domain. FLAC's domain is audio data. Though it can losslessly code any input, only certain kinds of input will get smaller. FLAC exploits the fact that audio data typically has a high degree of sample-to-sample correlation.
Within the audio domain, there are many possible subdomains. For example: low bitrate speech, high-bitrate multi-channel music, etc. FLAC itself does not target a specific subdomain but many of the default parameters of the reference encoder are tuned to CD-quality music data (i.e. 44.1kHz, 2 channel, 16 bits per sample). The effect of the encoding parameters on different kinds of audio data will be examined later.
Architecture
Similar to many audio coders, a FLAC encoder has the following stages:
Definitions
Many terms like "block" and "frame" are used to mean different things in differenct encoding schemes. For example, a frame in MP3 corresponds to many samples across several channels, whereas an S/PDIF frame represents just one sample for each channel. The definitions we use for FLAC follow. Note that when we talk about blocks and subblocks we are referring to the raw unencoded audio data that is the input to the encoder, and when we talk about frames and subframes, we are referring to the FLAC-encoded data.
The size used for blocking the audio data has a direct effect on the compression ratio. If the block size is too small, the resulting large number of frames mean that excess bits will be wasted on frame headers. If the block size is too large, the characteristics of the signal may vary so much that the encoder will be unable to find a good predictor. In order to simplify encoder/decoder design, FLAC imposes a minimum block size of 16 samples, and a maximum block size of 65535 samples. This range covers the optimal size for all of the audio data FLAC supports.
Currently the reference encoder uses a fixed block size, optimized on the sample rate of the input. Future versions may vary the block size depending on the characteristics of the signal.
Blocked data is passed to the predictor stage one subblock (channel) at a time. Each subblock is independently coded into a subframe, and the subframes are concatenated into a frame. Because each channel is coded separately, it means that one channel of a stereo frame may be encoded as a constant subframe, and the other an LPC subframe.
Interchannel Decorrelation
In stereo streams, many times there is an exploitable amount of correlation between the left and right channels. FLAC
For a user-oriented overview, see About the FLAC Format.
Table of Contents
- Acknowledgments
- Scope
- Architecture
- Definitions
- Blocking
- Interchannel Decorrelation
- Prediction
- Residual Coding
- Format
- FLAC Subset
- Specification
FLAC owes much to the many people who have advanced the audio compression field so freely. For instance:
- A. J. Robinson for his work onShorten; his paper is a good starting point on some of the basic methods used by FLAC. FLAC trivially extends and improves the fixed predictors, LPC coefficient quantization, and Rice coding used in Shorten.
- S. W. Golomb and Robert F. Rice; their universal codes are used by FLAC's entropy coder.
- N. Levinson and J. Durbin; the reference encoder uses an algorithm developed and refined by them for determining the LPC coefficients from the autocorrelation coefficients.
- And of course, Claude Shannon
It is a known fact that no algorithm can losslessly compress all possible input, so most compressors restrict themselves to a useful domain and try to work as well as possible within that domain. FLAC's domain is audio data. Though it can losslessly code any input, only certain kinds of input will get smaller. FLAC exploits the fact that audio data typically has a high degree of sample-to-sample correlation.
Within the audio domain, there are many possible subdomains. For example: low bitrate speech, high-bitrate multi-channel music, etc. FLAC itself does not target a specific subdomain but many of the default parameters of the reference encoder are tuned to CD-quality music data (i.e. 44.1kHz, 2 channel, 16 bits per sample). The effect of the encoding parameters on different kinds of audio data will be examined later.
Architecture
Similar to many audio coders, a FLAC encoder has the following stages:
- Blocking. The input is broken up into many contiguous blocks. With FLAC, the blocks may vary in size. The optimal size of the block is usually affected by many factors, including the sample rate, spectral characteristics over time, etc. Though FLAC allows the block size to vary within a stream, the reference encoder uses a fixed block size.
- Interchannel Decorrelation. In the case of stereo streams, the encoder will create mid and side signals based on the average and difference (respectively) of the left and right channels. The encoder will then pass the best form of the signal to the next stage.
- Prediction. The block is passed through a prediction stage where the encoder tries to find a mathematical description (usually an approximate one) of the signal. This description is typically much smaller than the raw signal itself. Since the methods of prediction are known to both the encoder and decoder, only the parameters of the predictor need be included in the compressed stream. FLAC currently uses four different classes of predictors (described in the prediction section), but the format has reserved space for additional methods. FLAC allows the class of predictor to change from block to block, or even within the channels of a block.
- Residual coding. If the predictor does not describe the signal exactly, the difference between the original signal and the predicted signal (called the error or residual signal) must be coded losslessy. If the predictor is effective, the residual signal will require fewer bits per sample than the original signal. FLAC currently uses only one method for encoding the residual (see theResidual coding section), but the format has reserved space for additional methods. FLAC allows the residual coding method to change from block to block, or even within the channels of a block.
Definitions
Many terms like "block" and "frame" are used to mean different things in differenct encoding schemes. For example, a frame in MP3 corresponds to many samples across several channels, whereas an S/PDIF frame represents just one sample for each channel. The definitions we use for FLAC follow. Note that when we talk about blocks and subblocks we are referring to the raw unencoded audio data that is the input to the encoder, and when we talk about frames and subframes, we are referring to the FLAC-encoded data.
- Block: One or more audio samples that span several channels.
- Subblock: One or more audio samples within a channel. So a block contains one subblock for each channel, and all subblocks contain the same number of samples.
- Blocksize: The number of samples in any of a block's subblocks. For example, a one second block sampled at 44.1KHz has a blocksize of 44100, regardless of the number of channels.
- Frame: A frame header plus one or more subframes.
- Subframe: A subframe header plus one or more encoded samples from a given channel. All subframes within a frame will contain the same number of samples.
The size used for blocking the audio data has a direct effect on the compression ratio. If the block size is too small, the resulting large number of frames mean that excess bits will be wasted on frame headers. If the block size is too large, the characteristics of the signal may vary so much that the encoder will be unable to find a good predictor. In order to simplify encoder/decoder design, FLAC imposes a minimum block size of 16 samples, and a maximum block size of 65535 samples. This range covers the optimal size for all of the audio data FLAC supports.
Currently the reference encoder uses a fixed block size, optimized on the sample rate of the input. Future versions may vary the block size depending on the characteristics of the signal.
Blocked data is passed to the predictor stage one subblock (channel) at a time. Each subblock is independently coded into a subframe, and the subframes are concatenated into a frame. Because each channel is coded separately, it means that one channel of a stereo frame may be encoded as a constant subframe, and the other an LPC subframe.
Interchannel Decorrelation
In stereo streams, many times there is an exploitable amount of correlation between the left and right channels. FLAC
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