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ReplayGain is a proposed technical standard published by David Robinson in 2001 to measure and normalize the perceived loudness of audio in computer audio formats such as MP3 and Ogg Vorbis. It allows media players to normalize loudness for individual tracks or albums.
FFmpeg is a free and open-source software project consisting of a suite of libraries and programs for handling video, audio, and other multimedia files and streams. At its core is the command-line ffmpeg tool itself, designed for processing video and audio files.
Increasing the audio volume can lead to distortion if, on playback, multiplying the audio data by the global gain may increase the volume past its clipping threshold. Audio recordings that use Compressed audio to cap the overall volume may have already adjusted the global gain field. This may limit the ability to further amplify the recording ...
FFmpeg (decoding only), [7] FFmpeg with VisualOn libraries, Android (decoding only) [8] voice recording, audio No No No Yes No G.723.1: ITU-T 1996-03 G.723.1 (05/06) Non-free Various proprietary VoIP software FFmpeg voice recording: No Yes No Yes No G.726: ITU-T 1990-12 Free Various proprietary VoIP software FFmpeg, Ekiga and other VoIP ...
OptimFROG is a proprietary, lossless audio codec developed by Florin Ghido. OptimFROG is optimized for high compression (small file sizes) at the expense of encoding and decoding speed, and consistently measures among the highest compressing lossless codecs.
Free and open-source software portal; libavcodec is a free and open-source [4] library of codecs for encoding and decoding video and audio data. [5]libavcodec is an integral part of many open-source multimedia applications and frameworks.
Flash Player clients, when recording audio from a user's microphone, can use the Nellymoser Asao codec. (Flash Player 10 released in 2008 also supports the open source Speex codec. [3]) The sampling rate of the audio capture can be controlled by the Flash programmer to increase and decrease encoding bitrate and quality.
Advanced Audio Coding (AAC) is an audio coding standard for lossy digital audio compression.It was designed to be the successor of the MP3 format and generally achieves higher sound quality than MP3 at the same bit rate.