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foobar2000 [a] (often abbreviated as fb2k or f2k) is a freeware audio player for Microsoft Windows, iOS, Android, macOS, and formerly Windows Phone, developed by Peter Pawłowski. It has a modular design , which provides user flexibility in configuration and customization. [ 4 ]
Pre-mixed PCM audio is sent to the driver in a single format (in terms of sample rate, bit depth and channel count) that is configurable from the Sounds control panel. WASAPI Exclusive mode bypasses the mixer, as does using third-party audio APIs like OpenAL or ASIO, which still have direct access to the hardware. [14]
Foo Camp is an annual hacker convention.; BarCamp, an international network of user-generated conferences.; During the United States v.Microsoft Corp. trial, evidence was presented that Microsoft had tried to use the Web Services Interoperability organization (WS-I) as a means to stifle competition, including e-mails in which top executives including Bill Gates and Steve Ballmer referred to ...
Audio Stream Input/Output (ASIO) is a computer audio interface driver protocol for digital audio specified by Steinberg, providing high data throughput, synchronization, and low latency between a software application and a computer's audio interface or sound card.
PortAudio is an open-source computer library for audio playback and recording. It is a cross-platform library, so programs using it can run on many different computer operating systems, including Windows, Mac OS X and Linux.
16 bit 40 ms Yes No Yes: only in MPEG-4 Part 12 container: Yes: only in MPEG-4 Part 12 container: G.721: ADPCM, Lossy: 8 kHz 32 kbit/s 13 bit Yes No No No G.722: sub-band ADPCM, Lossy: 16 kHz 64 kbit/s (comprises 48, 56 or 64 kbit/s audio and 16, 8 or 0 kbit/s auxiliary data) 14 bit 4 ms Yes No No No G.722.1
WavPack compression can compress (and losslessly restore) 8, 16, 24, and 32-bit fixed-point, and 32-bit floating-point PCM audio files in the .WAV file format. It can also handle DSD input in DSDIFF or DSF format. [2] It also supports surround sound streams and high sampling rates. Like other lossless compression schemes, the data reduction ...
The MPEG-4 audio coding algorithm family spans the range from low bit rate speech encoding (down to 2 kbit/s) to high-quality audio coding (at 64 kbit/s per channel and higher). AAC offers sampling frequencies between 8 kHz and 96 kHz and any number of channels between 1 and 48.