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.https://hydrogenaud.io/index.php?topic=80510.0
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.
A Mac OSX GUI frontend for open source audio codecs. There is an older Windows version available as well.
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.
FAAC is an Advanced Audio Coder (MPEG2-AAC, MPEG4-AAC). The goal of FAAC is to explore the possibilities of AAC and exceed the quality of the currently best MP3 encoders.
- High quality audio
- High-speed encoding
- LC, Main, LTP support
- DRM support through DreaM
WavPack is a completely open audio compression format providing lossless, high-quality lossy, and a unique hybrid compression mode. For version 5.0.0, several new file formats and lossless DSD audio compression were added, making WavPack a universal audio archiving solution.
In the default lossless mode WavPack acts just like a WinZip compressor for audio files. However, unlike MP3 or WMA encoding which can affect the sound quality, not a single bit of the original information is lost, so there’s no chance of degradation. This makes lossless mode ideal for archiving audio material or any other situation where quality is paramount. The compression ratio depends on the source material, but generally is between 30% and 70%.
The hybrid mode provides all the advantages of lossless compression with an additional bonus. Instead of creating a single file, this mode creates both a relatively small, high-quality lossy file that can be used all by itself, and a “correction” file that (when combined with the lossy file) provides full lossless restoration. For some users this means never having to choose between lossless and lossy compression!
WavPack employs only well known, public domain techniques (i.e., linear prediction with LMS adaptation, Elias and Golomb codes) in its implementation. Methods and algorithms that have ever been patented (e.g., arithmetic coding, LZW compression) are specifically avoided. This ensures that WavPack encoders and decoders will remain open and royalty-free.