What sets this frontend for ffmpeg apart from other stereo simulators is it creates an illusion of actual stereo separation. More importantly, it produces none of the weird phasing, and/or time delay artifacts. And very little, if any of the tone discoloration, when those others aren’t meticulously set up just right. All this is accomplished by using the ffmpeg crossover audio filter to split the sound into 8 frequency bands. The split points are based on center frequencies of a typical 1/3 octave equalizer. Those 8 bands are then panned in varying degrees to left and right. The varying width of each band is set to achieve the best balance between the left and right channels.
PS3 Media Server is a DLNA-compliant UPnP Media Server. Originally written to support the Sony PlayStation3, PS3 Media Server has been expanded to support a range of other media renderers, including smartphones, TVs, music players and more. Because it is written in Java, PS3 Media Server supports all major operating systems, with versions for Windows, Linux and Mac OS X.
The program streams or transcodes many different media formats with little or no configuration. It is powered by MEncoder, FFmpeg, tsMuxeR and AviSynth, which combine to offer support for a wide range of media formats.
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.
GNUsound is a multitrack sound editor for GNOME 1 and 2.
- Easy to use and clean user interface.
- Hide the advanced options with the ability to show them.
- Convert to more than 100 different formats.
- Allow to edit formats.
- Shutdown or suspend PC afer a conversion process.
- Show file informations (duration, remaining time, estimated size, progress value).
- Show file details using mediainfo.
- Allow to skip or remove file during conversion process.
- Preview file before conversion.
Dynamic Audio Normalizer is a library for advanced audio normalization purposes. It applies a certain amount of gain to the input audio in order to bring its peak magnitude to a target level (e.g. 0 dBFS). However, in contrast to more “simple” normalization algorithms, the Dynamic Audio Normalizer dynamically re-adjusts the gain factor to the input audio. This allows for applying extra gain to the “quiet” sections of the audio while avoiding distortions or clipping the “loud” sections. In other words: The Dynamic Audio Normalizer will “even out” the volume of quiet and loud sections, in the sense that the volume of each section is brought to the same target level. Note, however, that the Dynamic Audio Normalizer achieves this goal without applying “dynamic range compression”. It will retain 100% of the dynamic range within each “local” region of the audio file.
The Dynamic Audio Normalizer is available as a small standalone command-line utility and also as an effect in the SoX audio processor as well as in the FFmpeg audio/video converter. Furthermore, it can be integrated into your favourite DAW (digital audio workstation), as a VST plug-in, or into your favourite media player, as a Winamp plug-in. Last but not least, the “core” library can be integrated into custom applications easily, thanks to a straightforward API (application programming interface). The “native” API is written in C++, but language bindings for C99 Microsoft.NET, Java, Python and Pascal are provided.