TAK stands for (T)om’s lossless (A)audio (k)ompressor. Besides, it’s a throwback to a (not very philanthropic) character from Stephen King’s “Regulators”. Early semi-public evaluation versions operated under the working title YALAC .
High compression . The strongest mode is on a par with Monkey’s Audio High and OptimFrog’s Normal; for specific files such as classical music or voice recordings, it often outperforms both. This classification is based on the evaluation of hundreds of files of various styles; it definitely does not apply to every single file.
High compression speed . I am currently not aware of any other compressor that works faster than TAK’s Turbo or Fast mode and achieves similar compression rates.
Multi-core compressor . The compressor optionally generates up to four threads in order to take advantage of multi-core cpus.
Very high decompression speed . It is at the level of FLAC and therefore significantly higher than most symmetrical compressors.
Support for every popular audio format (not yet fully implemented).
Streaming support . An info frame, which contains all the information required for decoding, is inserted into the compressed audio data every 2 seconds.
Fault tolerance . A single bit error never damages the audio data for more than a maximum of 250 ms, since it is stored in completely independent frames of a maximum of this duration. The decoder processes even extremely damaged files, optionally replacing or removing the affected data with silence.
Error detection . Every single frame is protected by a 24-bit checksum (CRC).
MD5 checksums for quick identification of audio material (e.g. for searching for duplicates).
Fast, sample-accurate access to any playback position . The file header contains a look-up table with index positions every second. Even without this table, efficient random access is possible; for this purpose, the synchronization codes of the frame headers and / or the offset values optionally recorded in the frame header, which refer to the beginning of the previous and next frame, can be used.
Metadata . A flexible and expandable structure allows the recording of non-audio data such as images or cuesheets.
Playback plugins for Winamp and Foobar are currently available.
An SDK provides other developers with decoding functions for integration into their applications. An extension to include coding functions is planned.
qaac requires Apple Application Support that is included in iTunes or QuickTime. If you are using 64 bit Windows, recent 64 bit iTunes (ver 12.1 or later) is recommended. 64 bit installer for iTunes (iTunes6464Setup.exe) includes both of 32 bit and 64 bit Apple Application Support, therefore you can run both of qaac.exe and qaac64.exe with its installation.
AAC-LC, AAC-HE, ALAC encoding are supported. m4a container (just a MP4 container with ‘M4A ‘ brand, created by Apple) is used by default, but you can also mux into ADTS.
Filenames and tags are treated with Unicode. Very long file names are supported.
Support for multichannel / surround.
Fine quality control over AAC encoding.
Support for hi-resolution / multichannel ALAC.
WAV, raw PCM, ALAC, and all LPCM formats supported by CoreAudio AudioFile interface (such as AIFF, CAF, and Sun/AU) are directly available as input. MP3 is also decodable through CoreAudio.
Cue sheet input is also available.
FLAC, Wavpack, TAK, and other LPCM formats are optionally supported when libFLAC.dll, wavpackdll.dll, tak_deco_lib.dll, and libsndfile-1.dll are installed.
Piped input is available for WAV and raw PCM.
Piped output (streaming) is supported for ADTS.
Automatically fetch tags from AIFF, FLAC, Wavpack, Tak, and ALAC. Major tags are copied to the result. Also, you can manually set tags via command line options.
A complete, cross-platform solution to record, convert and stream audio and video. FFmpeg is the leading multimedia framework, able to decode, encode, transcode, mux, demux, stream, filter and play pretty much anything that humans and machines have created. It supports the most obscure ancient formats up to the cutting edge. No matter if they were designed by some standards committee, the community or a corporation. It is also highly portable: FFmpeg compiles, runs, and passes our testing infrastructure FATE across Linux, Mac OS X, Microsoft Windows, the BSDs, Solaris, etc. under a wide variety of build environments, machine architectures, and configurations.
GXLame is an MP3 encoder based off of LAME v3.98.4 and v3.99b0 which has been heavily optimized for high-quality, low-bitrate VBR encoding. It is similar in concept to other popular encoders at these bitrates such as some AAC codecs, Vorbis mods, and so forth at bitrates down to 56kbps. This codec does not rely on aggressive lowpassing or resampling to achieve these low bitrates, and the quality aims to be acceptable at much lower bitrates than have come to be expected of the standard.
exhale, which is an acronym for “Ecodis eXtended High-efficiency And Low-complexity Encoder”, is a lightweight library and application to encode uncompressed WAVE-format audio files into MPEG-4 format files complying with the ISO/IEC 23003-3 (MPEG-D) Unified Speech and Audio Coding (USAC, also known as Extended High-Efficiency AAC) standard. In addition, exhale writes program peak-level and loudness data into the generated MPEG-4 files according to the ISO/IEC 23003-4, Dynamic Range Control (DRC) specification for use by decoders providing DRC. exhale currently makes use of all frequency-domain (FD) coding tools in the scale factor based MDCT processing path, except for predictive joint stereo, which is still being integrated. Its objective is high quality mono, stereo, and multichannel coding at medium and high bit rates, so the lower-rate USAC coding tools (ACELP, TCX, Enhanced SBR and MPEG Surround with Unified Stereo coding) won’t be integrated.
Aften is an audio encoder which generates compressed audio streams based on ATSC A/52 specification. This type of audio is also known as AC-3 or Dolby® Digital and is one of the audio codecs used in DVD-Video content.
Converts PCM source audio into AC-3 audio
Allows for setting of production metadata such as dialogue level, downmix levels, and extended bitstream information
Supports RAW, WAV, AIFF, and CAFF input file formats
Supports single multi-channel or multiple mono source files
Includes several optional pre-processing filters
Uses multi-threading and assembly optimizations to speed up encoding
FLAC Frontend is a convenient way for Windows users not used to working with command lines to use the official FLAC tools. It accepts WAVE, W64, AIFF and RAW files for encoding and outputs FLAC or OGG-FLAC files. It is able to decode FLAC files, test them, fingerprint them and re-encode them. It has drag-and-drop support too. It is tested on Windows XP SP3 and Windows 7, but should work with Windows XP SP2 or newer. It requires .NET 2.0 or later.
TooLAME is a free software MPEG-1 Layer II (MP2) audio encoder written primarily by Mike Cheng. While there are many MP2 encoders, TooLAME is well-known and widely used for its particularly high audio quality. It has been unmaintained since 2003, but is directly succeeded by the TwoLAME code fork (the latest version, TwoLAME 0.3.13, was released January 21, 2011). The name TooLAME is a play on LAME and Layer II.