Wideband audio, also known as wideband voice or HD voice, is high definition voice quality for telephony audio, contrasted with standard digital telephony "toll quality". It extends the frequency range of audio signals transmitted over telephone lines, resulting in higher quality speech. The range of the human voice extends from 100 Hz to 17 kHz but traditional, voiceband or narrowband telephone calls limit audio frequencies to the range of 300 Hz to 3.4 kHz. Wideband audio relaxes the bandwidth limitation and transmits in the audio frequency range of 50 Hz to 7 kHz. In addition, some wideband codecs may use a higher audio bit depth of 16 bits to encode samples, also resulting in much better voice quality.
Wideband codecs have a typical sample rate of 16 kHz. For superwideband codecs the typical value is 32 kHz.
In 1987, the International Telecommunication Union (ITU) standardized a version of wideband audio known as G.722. Radio broadcasters began using G.722 over Integrated Services Digital Network (ISDN) to provide high-quality audio for remote broadcasts, such as commentary from sports venues. AMR-WB (G.722.2) was developed by Nokia and VoiceAge and it was first specified by 3GPP.
The traditional telephone network (PSTN) is generally limited to narrowband audio by the intrinsic nature of its transmission technology, TDM (time-division multiplexing), and by the analogue-to-digital converters used at the edge of the network, as well as the speakers, microphones and other elements in the endpoints themselves.
Wideband audio has been broadly deployed in conjunction with videoconferencing. Providers of this technology quickly discovered that despite the explicit emphasis on video transmission, the quality of the participant experience was significantly influenced by the fidelity of the associated audio signal.
Communications via Voice over Internet Protocol (VoIP) can readily employ wideband audio. When PC-to-PC calls are placed via VoIP services, such as Skype, and the participants use a high-quality headset, the resulting call quality can be noticeably superior to conventional PSTN calls.
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Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors. Opus replaces both Vorbis and Speex for new applications, and several blind listening tests have ranked it higher-quality than any other standard audio format at any given bitrate until transparency is reached, including MP3, AAC, and HE-AAC.
An audio coding format (or sometimes audio compression format) is a content representation format for storage or transmission of digital audio (such as in digital television, digital radio and in audio and video files). Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.
Speech coding is an application of data compression to digital audio signals containing speech. Speech coding uses speech-specific parameter estimation using audio signal processing techniques to model the speech signal, combined with generic data compression algorithms to represent the resulting modeled parameters in a compact bitstream. Common applications of speech coding are mobile telephony and voice over IP (VoIP).
The goal of this course is to introduce the engineering students state-of-the-art speech and audio coding techniques with an emphasis on the integration of knowledge about sound production and auditor
Explores computer-mediated communication tools, media richness, grounding theory, biases in communication, and managing virtual teams.
The growing requirements for broadcasting and streaming of high quality video continue to trigger demands for codecs with higher compression efficiency. AV1 is the most recent open and royalty free video coding specification developed by Alliance for Open ...
2018
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n-linear impedance characteristics of VO2 offer solutions to highly sensitive radio-frequency (RF), millimeterwave (mm-wave), and terahertz (THz) detectors. We demonstrate high-speed broadband sensibility of miniaturized VO2 switches by utilizing the nonli ...
IEEE2022
Subword modeling for zero-resource languages aims to learn low-level representations of speech audio without using transcriptions or other resources from the target language (such as text corpora or pronunciation dictionaries). A good representation should ...