Flac Changelog

[Baldomero Prado]

As there have been changes to the library interfaces, the libFLAC version number is incremented to 12, the libFLAC++ version number is incremented to 10. As some changes were breaking, the version age numbers (see libtool versioning) have been reset to 0. For more details on the changes to the API, see the porting guide.

This is the first beta version of FLAC. Being beta, there will be no changes to the format that will break older streams, unless a serious bug involving the format is found. What this means is that, barring such a bug, streams created with 0.5 will be decodable by future versions. This version also includes some new features:

This version fixes a bug in the constant subframe detection. More importantly, a verify option (-V) has been added to flac that verifies the encoding process. With this option turned on, flac will create a parallel decoder while encoding to make sure that the encoded output decodes to exactly match the original input. In this way, any unknown bug in the encoder will be caught and flac will abort with an error message.

The development for this release focused on the tools, flac and metaflac, whereas the previous few releases had the focus on libFLAC. The main improvement in flac is its handling of foreign metadata on decoding a file (i.e. restoring the foreign metadata), metaflac has had a few features added. Both tools have been vetted with fuzzing, resulting in numerous small fixes.

This release fixes a lot of bugs found by fuzzing, and fuzzing coverage has improved much. Also, compression should be better but a little slower. Speedups for 64-bit ARM have been implemented. Encoding and decoding 32 bps PCM is now possible.

When building with MSVC, using CMake is preferred, see the README under

"Building with CMake" for more information. Building with MSVC using

solution files is deprecated and these files will be removed in the future.

FLAC (/flk/; Free Lossless Audio Codec) is an audio coding format for lossless compression of digital audio, developed by the Xiph.Org Foundation, and is also the name of the free software project producing the FLAC tools, the reference software package that includes a codec implementation. Digital audio compressed by FLAC's algorithm can typically be reduced to between 50 and 70 percent of its original size[4] and decompresses to an identical copy of the original audio data.

Development was started in 2000 by Josh Coalson. The bitstream format was frozen with the release of version 0.9 of the reference implementation on 31 March 2001. Version 1.0 was released on 20 July 2001.[5]

On 29 January 2003, the Xiph.Org Foundation and the FLAC project announced the incorporation of FLAC under the Xiph.org banner. Xiph.org is home to other free compression formats such as Vorbis, Theora, Speex and Opus.[5][6][7]

The encoded audio is divided into frames, each of which consists of a header, a data block, and a CRC16 checksum. Each frame is encoded independent of each other. A frame header begins with a sync word, used to identify the beginning of a valid frame. The rest of the header contains the number of samples, position of the frame, channel assignment, and optionally the sample rate and bit depth. The data block contains the audio information.[10]

Metadata in FLAC precedes the audio. Properties like the sample rate and the number of channels are always contained in the metadata. It may also contain other information, the album cover for example.[10] FLAC uses Vorbis comments for textual metadata like track title and artist name.

The FLAC encoding algorithm consists of multiple stages. In the first stage, the input audio is split into blocks. If the audio contains multiple channels, each channel is encoded separately as a subblock. The encoder then tries to find a good mathematical approximation of the block, either by fitting a simple polynomial, or through general linear predictive coding. A description of the approximation, which is only a few bytes in length, is then written. Finally, the difference between the approximation and the input, called residual, is encoded using Rice coding. In many cases, a description of the approximation and the encoded residual takes up less space than using pulse-code modulation.[10]

The decoding process is the reverse of encoding. The compressed residual is first decoded.[11][12] The description of the mathematical approximation is then used to calculate a waveform. The result is formed by adding the residual and the calculated waveform.[13][14] As FLAC compresses losslessly, the decoded waveform is identical to the waveform before encoding.

For two-channel stereo, the encoder may choose to joint-encode the audio. The channels are transformed into a side channel, which is the difference between the two input channels, and a mid channel, the sum of the two input channels. In place of a mid channel, the left channel or the right channel may be encoded instead, which is sometimes more space-efficient.[15]

The amount of compression is determined by various parameters, including the order of the linear prediction model and the block size. Regardless of the amount of compression, the original data can always be reconstructed perfectly.

For user's convenience, the reference implementation defines 9 compression levels, which are presets of the more technical parameters to the encoding algorithm. The levels are labeled from 0 to 8, with higher numbers resulting in a higher compression ratio, at the cost of compression speed. The meaning of each compression level varies by implementation.[16][17]

FLAC is optimized for decoding speed at the expense of encoding speed. A benchmark has shown that, while there is little variation in decoding speed as compression level increases, beyond the default compression level 5, the encoding process takes up considerably more time with little space saved compared to level 5.[18]

Alongside the format, the FLAC project also contains a free and open-source reference implementation of FLAC called libFLAC. libFLAC contains facilities to encode and decode FLAC data and to manipulate the metadata of FLAC files. libFLAC++, an object-oriented wrapper around libFLAC for C++, and the command-line programs flac and metaflac, are also part of the reference implementation.

Since FLAC is a lossless scheme, it is suitable as an archive format for owners of CDs and other media who wish to preserve their audio collections. If the original media are lost, damaged, or worn out, a FLAC copy of the audio tracks ensures that an exact duplicate of the original data can be recovered at any time. An exact restoration from a lossy copy (e.g., MP3) of the same data is impossible. FLAC's being lossless means it is highly suitable for transcoding e.g. to MP3, without the normally associated transcoding quality loss between one lossy format and another. A CUE file can optionally be created when ripping a CD. If a CD is read and ripped perfectly to FLAC files, the CUE file allows later burning of an audio CD that is identical in audio data to the original CD, including track order and pregap, but excluding additional data such as lyrics and CD+G graphics.[19] But depending on the burning program used, CD-Text may be recovered from the metadata stored in the CUE sheet and burned back to a new copy on blank CD-R media.

The reference implementation of FLAC is implemented as the libFLAC core encoder & decoder library, with the main distributable program flac being the reference implementation of the libFLAC API. This codec API is also available in C++ as libFLAC++. The reference implementation of FLAC compiles on many platforms, including most Unix (such as Solaris, BSD) and Unix-like (including Linux), Microsoft Windows, BeOS, and OS/2 operating systems. There are build-systems for autoconf/automake, MSVC, Watcom C, and Xcode. There is currently no multicore support in libFLAC, but utilities such as GNU parallel and various graphical frontends can be used to spin up multiple instances of the encoder.

FLAC playback support in portable audio devices and dedicated audio systems is limited compared to formats such as MP3[20] or uncompressed PCM. FLAC support is included by default in Windows 10, Android, BlackBerry 10 and Jolla devices.

In 2014, several aftermarket mobile electronics companies introduced multimedia solutions that include support for FLAC. These include the NEX series from Pioneer Electronics and the VX404 and NX404 from Clarion.

The European Broadcasting Union (EBU) has adopted the FLAC format for the distribution of high quality audio over its Euroradio network.[21] The Windows operating system has supported native FLAC integration since the introduction of Windows 10.[22] The Android operating system has supported native FLAC playback since version 3.1.[23][24] macOS High Sierra and iOS 11 add native FLAC playback support.[25]

Among others the Pono music player and streaming service used the FLAC format.[26][27] Bandcamp insists on a lossless format for uploading, and has FLAC as a download option.[28] The Wikimedia Foundation sponsored a free and open-source online ECMAScript FLAC tool for browsers supporting the required HTML5 features.[29]

Name : flacProduct : Fedora 36Version : 1.3.4Release : 1.fc36URL : : An encoder/decoder for the Free Lossless Audio CodecDescription :FLAC stands for Free Lossless Audio Codec. Grossly oversimplified, FLACis similar to Ogg Vorbis, but lossless. The FLAC project consists ofthe stream format, reference encoders and decoders in library form,flac, a command-line program to encode and decode FLAC files, metaflac,a command-line metadata editor for FLAC files and input plugins forvarious music players.

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