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39-551: Algebraic code-excited linear prediction ( ACELP ) is a speech coding algorithm in which a limited set of pulses is distributed as excitation to a linear prediction filter. It is a linear predictive coding (LPC) algorithm that is based on the code-excited linear prediction (CELP) method and has an algebraic structure. ACELP was developed in 1989 by the researchers at the Université de Sherbrooke in Canada . The ACELP method

78-446: A 500 mile distance, on an average US network conditions. Latency is a larger consideration when an echo is present and systems must perform echo suppression and cancellation . Latency can be a particular problem in audio platforms on computers. Supported interface optimizations reduce the delay down to times that are too short for the human ear to detect. By reducing buffer sizes, latency can be reduced. A popular optimization solution

117-651: A bitrate of 64 kbit/s is the encoding method predominantly used on the public switched telephone network . The AMR narrowband codec, used in GSM and UMTS networks, introduces latency in the encode and decode processes. As mobile operators upgrade existing best-effort networks to support concurrent multiple types of service over all-IP networks, services such as Hierarchical Quality of Service ( H-QoS ) allow for per-user, per-service QoS policies to prioritise time-sensitive protocols like voice calls, and other wireless backhaul traffic. Another aspect of mobile latency

156-423: A distance from the stage but closer to the rear of the audience. Sound travels through air at the speed of sound (around 343 metres (1,125 ft) per second depending on air temperature and humidity). By measuring or estimating the difference in latency between the loudspeakers near the stage and the loudspeakers nearer the audience, the audio engineer can introduce an appropriate delay in the audio signal going to

195-418: A group can be from one another. Stage monitoring extends that limit, as sound travels close to the speed of light through the cables that connect stage monitors. Performers, particularly in large spaces, will also hear reverberation , or echo of their music, as the sound that projects from stage bounces off of walls and structures, and returns with latency and distortion. A primary purpose of stage monitoring

234-408: A higher tunable bitrate and is wideband. Latency (audio) Latency can be a critical performance metric in professional audio including sound reinforcement systems , foldback systems (especially those using in-ear monitors ) live radio and television . Excessive audio latency has the potential to degrade call quality in telecommunications applications. Low latency audio in computers

273-507: A low-amplitude noise is heard along a low-amplitude speech signal but is masked by a high-amplitude one. Although this would generate unacceptable distortion in a music signal, the peaky nature of speech waveforms, combined with the simple frequency structure of speech as a periodic waveform having a single fundamental frequency with occasional added noise bursts, make these very simple instantaneous compression algorithms acceptable for speech. A wide variety of other algorithms were tried at

312-426: A scalable structure, was standardized by ITU-T. The input sampling rate is 16 kHz. Much of the later work in speech compression was motivated by military research into digital communications for secure military radios , where very low data rates were used to achieve effective operation in a hostile radio environment. At the same time, far more processing power was available, in the form of VLSI circuits , than

351-610: A small amount of time to accomplish; typical latencies are in the range of 0.2 to 1.5 milliseconds, depending on sampling rate, software design and hardware architecture. Different audio signal processing operations such as finite impulse response (FIR) and infinite impulse response (IIR) filters take different mathematical approaches to the same end and can have different latencies. In addition, input and output sample buffering add delay. Typical latencies range from 0.5 to ten milliseconds with some designs having as much as 30 milliseconds of delay. Latency in digital audio equipment

390-452: A stable connection with sufficient bandwidth and minimal latency, VoIP systems typically have a minimum of 20 ms inherent latency. Under less ideal network conditions a 150 ms maximum latency is sought for general consumer use. Many popular videoconferencing systems rely on data buffering and data redundancy to cope for network jitter and packet loss. Measurements have shown that mouth-to-ear delay are between 160 to 300 ms over

429-553: Is linear predictive coding (LPC), while the most widely used in VoIP applications are the LPC and modified discrete cosine transform (MDCT) techniques. The techniques employed in speech coding are similar to those used in audio data compression and audio coding where appreciation of psychoacoustics is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in

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468-462: Is Steinberg's ASIO , which bypasses the audio platform, and connects audio signals directly to the sound card's hardware. Many professional and semi-professional audio applications utilize the ASIO driver, allowing users to work with audio in real time. Pro Tools HD offers a low latency system similar to ASIO. Pro Tools 10 and 11 are also compatible with ASIO interface drivers. The Linux realtime kernel

507-484: Is a modified kernel, that alters the standard timer frequency the Linux kernel uses and gives all processes or threads the ability to have realtime priority. This means that a time-critical process like an audio stream can get priority over another, less-critical process like network activity. This is also configurable per user (for example, the processes of user "tux" could have priority over processes of user "nobody" or over

546-549: Is given by: Speech coding 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 most widely used speech coding technique in mobile telephony

585-470: Is important for interactivity . In all systems, latency can be said to consist of three elements: codec delay, playout delay and network delay. Latency in telephone calls is sometimes referred to as mouth-to-ear delay ; the telecommunications industry also uses the term quality of experience (QoE). Voice quality is measured according to the ITU model; measurable quality of a call degrades rapidly where

624-532: Is most noticeable when a singer's voice is transmitted through their microphone, through digital audio mixing, processing and routing paths, then sent to their own ears via in-ear monitors or headphones. In this case, the singer's vocal sound is conducted to their own ear through the bones of the head, then through the digital pathway to their ears some milliseconds later. In one study, listeners found latency greater than 15 ms to be noticeable. Latency for other musical activities such as playing guitar does not have

663-457: Is often necessary to use channel coding for transmission, to avoid losses due to transmission errors. In order to get the best overall coding results, speech coding and channel coding methods are chosen in pairs, with the more important bits in the speech data stream protected by more robust channel coding. The modified discrete cosine transform (MDCT) is used in the LD-MDCT technique used by

702-470: Is required to succeed. Most of these games have a lag calibration setting whereupon the game will adjust the timing windows by a certain number of milliseconds to compensate. In these cases, the notes of a song will be sent to the speakers before the game even receives the required input from the player in order to maintain the illusion of rhythm. Games that rely upon musical improvisation , such as Rock Band drums or DJ Hero , can still suffer tremendously, as

741-414: Is the inter-network handoff; as a customer on Network A calls a Network B customer the call must traverse two separate Radio Access Networks , two core networks, and an interlinking Gateway Mobile Switching Centre (GMSC) which performs the physical interconnecting between the two providers. With end-to-end QoS managed and assured rate connections, latency can be reduced to analogue PSTN/POTS levels. On

780-412: Is the target for audio circuits within professional production structures. Latency in live performance occurs naturally from the speed of sound . It takes sound about 3 milliseconds to travel 1 meter. Small amounts of latency occur between performers depending on how they are spaced from each other and from stage monitors if these are used. This creates a practical limit to how far apart the artists in

819-669: Is to provide artists with more primary sound so that they are not confused by the latency of these reverberations. While analog audio equipment has no appreciable latency, digital audio equipment has latency associated with two general processes: conversion from one format to another, and digital signal processing (DSP) tasks such as equalization, compression and routing. Digital conversion processes include analog-to-digital converters (ADC), digital-to-analog converters (DAC), and various changes from one digital format to another, such as AES3 which carries low-voltage electrical signals to ADAT , an optical transport. Any such process takes

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858-401: Is used for example in the GSM standard. In CELP, the modeling is divided in two stages, a linear predictive stage that models the spectral envelope and a code-book-based model of the residual of the linear predictive model. In CELP, linear prediction coefficients (LPC) are computed and quantized, usually as line spectral pairs (LSPs). In addition to the actual speech coding of the signal, it

897-478: Is widely employed in current speech coding standards such as AMR , EFR , AMR-WB (G.722.2), VMR-WB , EVRC , EVRC-B , SMV , TETRA , PCS 1900, MPEG-4 CELP and ITU-T G-series standards G.729 , G.729.1 (first coding stage) and G.723.1 . The ACELP algorithm is also used in the proprietary ACELP.net codec. Audible Inc. use a modified version for their speaking books. It is also used in conference-calling software, speech compression tools and has become one of

936-629: Is widely used for VoIP calls in WhatsApp . The PlayStation 4 video game console also uses Opus for its PlayStation Network system party chat. A number of codecs with even lower bit rates have been demonstrated. Codec2 , which operates at bit rates as low as 450 bit/s, sees use in amateur radio. NATO currently uses MELPe , offering intelligible speech at 600 bit/s and below. Neural vocoder approaches have also emerged: Lyra by Google gives an "almost eerie" quality at 3 kbit/s. Microsoft's Satin also uses machine learning, but uses

975-479: The 3GPP formats. The ACELP patent expired in 2018 and is now royalty-free. The main advantage of ACELP is that the algebraic codebook it uses can be made very large (> 50 bits) without running into storage ( RAM / ROM ) or complexity ( CPU time) problems. The ACELP algorithm is based on that used in code-excited linear prediction (CELP), but ACELP codebooks have a specific algebraic structure imposed upon them. A 16-bit algebraic codebook shall be used in

1014-608: The AAC-LD format introduced in 1999. MDCT has since been widely adopted in voice-over-IP (VoIP) applications, such as the G.729.1 wideband audio codec introduced in 2006, Apple 's FaceTime (using AAC-LD) introduced in 2010, and the CELT codec introduced in 2011. Opus is a free software audio coder. It combines the speech-oriented LPC-based SILK algorithm and the lower-latency MDCT-based CELT algorithm, switching between or combining them as needed for maximal efficiency. It

1053-444: The G.114 recommendation regarding mouth-to-ear delay indicates that most users are "very satisfied" as long as latency does not exceed 200 ms, with an according R of 90+. Codec choice also plays an important role; the highest quality (and highest bandwidth) codecs like G.711 are usually configured to incur the least encode-decode latency, so on a network with sufficient throughput sub-100 ms latencies can be achieved. G.711 at

1092-411: The frequency band 400 to 3500 Hz is transmitted but the reconstructed signal retains adequate intelligibility . Speech coding differs from other forms of audio coding in that speech is a simpler signal than other audio signals, and statistical information is available about the properties of speech. As a result, some auditory information that is relevant in general audio coding can be unnecessary in

1131-407: The game cannot predict what the player will hit in these cases, and excessive lag will still create a noticeable delay between hitting notes, and hearing them play. Audio latency can be experienced in broadcast systems where someone is contributing to a live broadcast over a satellite or similar link with high delay. The person in the main studio has to wait for the contributor at the other end of

1170-440: The innovative codebook search, the aim of which is to find the best innovation and gain parameters. The innovation vector contains, at most, four non-zero pulses. In ACELP, a block of N speech samples is synthesized by filtering an appropriate innovation sequence from a codebook, scaled by a gain factor g c , through two time-varying filters. The long-term (pitch) synthesis filter is given by: The short-term synthesis filter

1209-486: The latter loudspeakers, so that the wavefronts from near and far loudspeakers arrive at the same time. Because of the Haas effect an additional 15 milliseconds can be added to the delay time of the loudspeakers nearer the audience, so that the stage's wavefront reaches them first, to focus the audience's attention on the stage rather than the local loudspeaker. The slightly later sound from delayed loudspeakers simply increases

Algebraic code-excited linear prediction - Misplaced Pages Continue

1248-494: The link to react to questions. Latency in this context could be between several hundred milliseconds and a few seconds. Dealing with audio latencies as high as this takes special training in order to make the resulting combined audio output reasonably acceptable to the listeners. Wherever practical, it is important to try to keep live production audio latency low in order to keep the reactions and interchange of participants as natural as possible. A latency of 10 milliseconds or better

1287-521: The mouth-to-ear delay latency exceeds 200 milliseconds. The mean opinion score (MOS) is also comparable in a near-linear fashion with the ITU's quality scale - defined in standards G.107, G.108 and G.109 - with a quality factor R ranging from 0 to 100. An MOS of 4 ('Good') would have an R score of 80 or above; to achieve 100R requires an MOS exceeding 4.5. The ITU and 3GPP groups end-user services into classes based on latency sensitivity: Similarly,

1326-433: The processes of several system daemons ). Many modern digital television receivers, set-top boxes and AV receivers use sophisticated audio processing, which can create a delay between the time when the audio signal is received and the time when it is heard on the speakers. Since TVs also introduce delays in processing the video signal this can result in the two signals being sufficiently synchronized to be unnoticeable by

1365-432: The same critical concern. Ten milliseconds of latency isn't as noticeable to a listener who is not hearing his or her own voice. In sound reinforcement for music or speech presentation in large venues, it is optimal to deliver sufficient sound volume to the back of the venue without resorting to excessive sound volumes near the front. One way for audio engineers to achieve this is to use additional loudspeakers placed at

1404-611: The speech coding context. Speech coding stresses the preservation of intelligibility and pleasantness of speech while using a constrained amount of transmitted data. In addition, most speech applications require low coding delay, as latency interferes with speech interaction. Speech coders are of two classes: The A-law and μ-law algorithms used in G.711 PCM digital telephony can be seen as an earlier precursor of speech encoding, requiring only 8 bits per sample but giving effectively 12 bits of resolution . Logarithmic companding are consistent with human hearing perception in that

1443-509: The time, mostly delta modulation variants, but after careful consideration, the A-law/μ-law algorithms were chosen by the designers of the early digital telephony systems. At the time of their design, their 33% bandwidth reduction for a very low complexity made an excellent engineering compromise. Their audio performance remains acceptable, and there was no need to replace them in the stationary phone network. In 2008, G.711.1 codec, which has

1482-414: The viewer. However, if the difference between the audio and video delay is significant, the effect can be disconcerting. Some systems have a lip sync setting that allows the audio lag to be adjusted to synchronize with the video, and others may have advanced settings where some of the audio processing steps can be turned off. Audio lag is also a significant detriment in rhythm games , where precise timing

1521-488: Was available for earlier compression techniques. As a result, modern speech compression algorithms could use far more complex techniques than were available in the 1960s to achieve far higher compression ratios. The most widely used speech coding algorithms are based on linear predictive coding (LPC). In particular, the most common speech coding scheme is the LPC-based code-excited linear prediction (CELP) coding, which

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