Ensonido

Last updated

Ensonido is a real-time post processing algorithm that allows users to play back MP3 Surround files in standard headphones. Ensonido was developed by the Fraunhofer Society. It simulates the natural reception of surround sound by the human ear, which usually receives tones from surrounding loudspeakers and from reflections and echoes of the listening room. The out-of-head localization achieved that way increases the listening comfort noticeably in contrast to conventional stereo headphone listening with its in-head localization of all sounds. In version 3.0 of the Fraunhofer IIS MP3 Surround Player, Ensonido is replaced with newer mp3HD.

Related Research Articles

<span class="mw-page-title-main">LAME</span> MPEG-1 Audio Layer 3 (MP3) audio encoder

LAME is a software encoder that converts digital audio into the MP3 audio coding format. LAME is a free software project that was first released in 1998 and has incorporated many improvements since then, including an improved psychoacoustic model. The LAME encoder outperforms early encoders like L3enc and possibly the "gold standard encoder" MP3enc, both marketed by Fraunhofer.

<span class="mw-page-title-main">MP3</span> Digital audio format

MP3 is a coding format for digital audio developed largely by the Fraunhofer Society in Germany under the lead of Karlheinz Brandenburg, with support from other digital scientists in other countries. Originally defined as the third audio format of the MPEG-1 standard, it was retained and further extended—defining additional bit rates and support for more audio channels—as the third audio format of the subsequent MPEG-2 standard. A third version, known as MPEG-2.5—extended to better support lower bit rates—is commonly implemented but is not a recognized standard.

<span class="mw-page-title-main">Vorbis</span> Royalty-free lossy audio encoding format

Vorbis is a free and open-source software project headed by the Xiph.Org Foundation. The project produces an audio coding format and software reference encoder/decoder (codec) for lossy audio compression, libvorbis. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis.

<span class="mw-page-title-main">Binaural recording</span> Method of recording sound

Binaural recording is a method of recording sound that uses two microphones, arranged with the intent to create a 3D stereo sound sensation for the listener of actually being in the room with the performers or instruments. This effect is often created using a technique known as dummy head recording, wherein a mannequin head is fitted with a microphone in each ear. Binaural recording is intended for replay using headphones and will not translate properly over stereo speakers. This idea of a three-dimensional or "internal" form of sound has also translated into useful advancement of technology in many things such as stethoscopes creating "in-head" acoustics and IMAX movies being able to create a three-dimensional acoustic experience.

<span class="mw-page-title-main">Head-related transfer function</span> Response that characterizes how an ear receives a sound from a point in space

A head-related transfer function (HRTF) is a response that characterizes how an ear receives a sound from a point in space. As sound strikes the listener, the size and shape of the head, ears, ear canal, density of the head, size and shape of nasal and oral cavities, all transform the sound and affect how it is perceived, boosting some frequencies and attenuating others. Generally speaking, the HRTF boosts frequencies from 2–5 kHz with a primary resonance of +17 dB at 2,700 Hz. But the response curve is more complex than a single bump, affects a broad frequency spectrum, and varies significantly from person to person.

<span class="mw-page-title-main">Headphones</span> Device placed near the ears that plays sound

Headphones are a pair of small loudspeaker drivers worn on or around the head over a user's ears. They are electroacoustic transducers, which convert an electrical signal to a corresponding sound. Headphones let a single user listen to an audio source privately, in contrast to a loudspeaker, which emits sound into the open air for anyone nearby to hear. Headphones are also known as earphones or, colloquially, cans. Circumaural and supra-aural headphones use a band over the top of the head to hold the drivers in place. Another type, known as earbuds or earpieces, consists of individual units that plug into the user's ear canal. A third type are bone conduction headphones, which typically wrap around the back of the head and rest in front of the ear canal, leaving the ear canal open. In the context of telecommunication, a headset is a combination of a headphone and microphone.

<span class="mw-page-title-main">WinDVD</span> Video and music player software

WinDVD is a commercial DVD video player software for Microsoft Windows.

<span class="mw-page-title-main">Adobe Audition</span> Digital audio workstation

Adobe Audition is a digital audio workstation developed by Adobe Inc. featuring both a multitrack, non-destructive mix/edit environment and a destructive-approach waveform editing view.

<span class="mw-page-title-main">MP3 Surround</span>

MP3 Surround is an extension of MP3 for multi-channel audio support including 5.1 surround sound. It was developed by Fraunhofer IIS in collaboration with Thomson and Agere Systems, and released in December 2004.

<span class="mw-page-title-main">Portable media player</span> Portable device capable of storing and playing digital media

A portable media player (PMP) or digital audio player (DAP) is a portable consumer electronics device capable of storing and playing digital media such as audio, images, and video files. The data is typically stored on a compact disc (CD), Digital Versatile Disc (DVD), Blu-ray Disc (BD), flash memory, microdrive, SD cards or hard drive; most earlier PMPs used physical media, but modern players mostly use flash memory. In contrast, analogue portable audio players play music from non-digital media that use analogue media, such as cassette tapes or vinyl records.

Fraunhofer l3enc was the first public software able to encode pulse-code modulation (PCM) .wav files to the MP3 format. The first public version was released on July 13, 1994. This command-line tool was shareware and limited to 112 kbit/s. It was available for MS-DOS, Linux, Solaris, SunOS, NeXTstep and IRIX. A licence that allowed full use cost 350 Deutsche Mark, or about $250 (US).

mp3 SX is a program that allows users to upgrade mp3 stereo files to MP3 Surround files. mp3 SX analyzes the existing natural ambience of the stereo material and plays it back through the rear channels. The sound sources remain in the front channels, but are played back through the left, center, and right channel, providing a stable front image even for off-sweet-spot listening. The mp3 SX program preserves the original stereo sound stage, creating additional surround envelopment, with only 15 kB/s additional information.

Sound Retrieval System (SRS) is a patented psychoacoustic 3D audio processing technology originally invented by Arnold Klayman in the early 1980s. The SRS technology applies head-related transfer functions (HRTFs) to create an immersive 3D soundfield using only two speakers, widening the "sweet spot", creating a more spacious sense of ambience, and producing strong localization cues for discrete instruments within an audio mix. SRS is not a Dolby matrix surround decoder but works with normal stereo recordings.

MPEG Surround, also known as Spatial Audio Coding (SAC) is a lossy compression format for surround sound that provides a method for extending mono or stereo audio services to multi-channel audio in a backwards compatible fashion. The total bit rates used for the core and the MPEG Surround data are typically only slightly higher than the bit rates used for coding of the core. MPEG Surround adds a side-information stream to the core bit stream, containing spatial image data. Legacy stereo playback systems will ignore this side-information while players supporting MPEG Surround decoding will output the reconstructed multi-channel audio.

Ambiophonics is a method in the public domain that employs digital signal processing (DSP) and two loudspeakers directly in front of the listener in order to improve reproduction of stereophonic and 5.1 surround sound for music, movies, and games in home theaters, gaming PCs, workstations, or studio monitoring applications. First implemented using mechanical means in 1986, today a number of hardware and VST plug-in makers offer Ambiophonic DSP. Ambiophonics eliminates crosstalk inherent in the conventional stereo triangle speaker placement, and thereby generates a speaker-binaural soundfield that emulates headphone-binaural sound, and creates for the listener improved perception of reality of recorded auditory scenes. A second speaker pair can be added in back in order to enable 360° surround sound reproduction. Additional surround speakers may be used for hall ambience, including height, if desired.

InfoZoom software is a data analysis, business intelligence and data visualization software product created using in-memory analytics. The software is created and supported by humanIT and the Fraunhofer Institute FIT, the same organization that created MP3 compression technology. The software has over 100,000 licensed users and over 1000 customers worldwide.

In the software development process, a reference implementation is a program that implements all requirements from a corresponding specification. The reference implementation often accompanies a technical standard, and demonstrates what should be considered the "correct" behavior of any other implementation of it.

Unified Speech and Audio Coding (USAC) is an audio compression format and codec for both music and speech or any mix of speech and audio using very low bit rates between 12 and 64 kbit/s. It was developed by Moving Picture Experts Group (MPEG) and was published as an international standard ISO/IEC 23003-3 and also as an MPEG-4 Audio Object Type in ISO/IEC 14496-3:2009/Amd 3 in 2012.

Fraunhofer FDK AAC is an open-source library for encoding and decoding digital audio in the Advanced Audio Coding (AAC) format. Fraunhofer IIS, developed this library for Android 4.1. It supports several Audio Object Types including MPEG-2 and MPEG-4 AAC LC, HE-AAC, HE-AACv2 as well AAC-LD and AAC-ELD for real-time communication. The encoding library supports sample rates up to 96 kHz and up to eight channels.

MPEG-H 3D Audio, specified as ISO/IEC 23008-3, is an audio coding standard developed by the ISO/IEC Moving Picture Experts Group (MPEG) to support coding audio as audio channels, audio objects, or higher order ambisonics (HOA). MPEG-H 3D Audio can support up to 64 loudspeaker channels and 128 codec core channels.