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30-538: Qualcomm code-excited linear prediction ( QCELP ), also known as Qualcomm PureVoice , is a speech codec developed in 1994 by Qualcomm to increase the speech quality of the IS-96A codec earlier used in CDMA networks. It was later replaced with EVRC since it provides better speech quality with fewer bits. The two versions, QCELP8 and QCELP13 , operate at 8 and 13 kilobits per second (Kbit/s) respectively. In CDMA systems,

60-414: A QCELP vocoder converts a sound signal into a signal transmissible within a circuit. In wired systems, voice signals are generally sampled at 8 kHz (that is, 8,000 sample values per second) and then encoded by 8-bit quantization for each sample value. Such a system transmits at 64 kbit/s, an expensive rate in a wireless system. A QCELP vocoder with variable rates can reduce the rate enough to fit

90-402: A comparator referenced to 0 (two levels quantizer), whose output is 1 or -1 if the quantizer's input is positive or negative. The demodulator is simply an integrator (like the one in the feedback loop) whose output rises or falls with each 1 or -1 received. The integrator itself constitutes a low-pass filter . The two sources of noise in delta modulation are "slope overload", when step size

120-490: A higher tunable bitrate and is wideband. Delta modulation To achieve high signal-to-noise ratio , delta modulation must use oversampling techniques, that is, the analog signal is sampled at a rate several times higher than the Nyquist rate . Derived forms of delta modulation are continuously variable slope delta modulation , delta-sigma modulation , and differential modulation . Differential pulse-code modulation

150-507: 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

180-538: A low-pass filter. ADM provides robust performance in the presence of bit errors meaning error detection and correction are not typically used in an ADM radio design, it is this very useful technique that allows for adaptive-delta-modulation. Contemporary applications of delta modulation includes, but is not limited to, recreating legacy synthesizer waveforms. With the increasing availability of FPGAs and game-related ASICs , sample rates are easily controlled so as to avoid slope overload and granularity issues. For example,

210-426: 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

240-449: A wireless system by coding the information more efficiently. In particular, it can change its own coding rates based on the speaker's volume or pitch; a louder or higher-pitched voice requires a higher rate. Speech codec 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

270-417: Is a modification of DM in which the step size is not fixed. Rather, when several consecutive bits have the same direction value, the encoder and decoder assume that slope overload is occurring, and the step size becomes progressively larger. Otherwise, the step size becomes gradually smaller over time. ADM reduces slope error, at the expense of increasing quantization error . This error can be reduced by using

300-758: Is available about the properties of speech. As a result, some auditory information that is relevant in general audio coding can be unnecessary in 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

330-452: 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

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360-455: Is the superset of DM. Rather than quantizing the value of the input analog waveform, delta modulation quantizes the difference between the current and the previous step, as shown in the block diagram in Fig. 1. The modulator is made by a quantizer which converts the difference between the input signal and the integral of the previous steps. In its simplest form, the quantizer can be realized with

390-433: Is too small to track the original waveform, and "granularity", when step size is too large. But a 1971 study shows that slope overload is less objectionable compared to granularity than one might expect based solely on SNR measures. In delta modulation, there is no limit to the number of pulses of the same sign that may occur, so it is capable of tracking slow-changing signals of any amplitude without clipping . However, if

420-446: 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

450-404: Is used to transmit only data that is relevant to the human auditory system. For example, in voiceband speech coding, only information in 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

480-624: 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

510-600: 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

540-647: The C64DTV used a 32 MHz sample rate, providing ample dynamic range to recreate the SID output to acceptable levels. Delta modulation was used by Satellite Business Systems (SBS) for its voice ports to provide long distance phone service to large domestic corporations with a significant inter-corporation communications need (such as IBM ). This system was in service throughout the 1980s. The voice ports used digitally implemented 24 kbit/s delta modulation with Voice Activity Compression (VAC) and echo suppressors to control

570-494: The fast gain error recovery increased the noise as determined by actual listening tests as compared to simple signal to noise measurements. The final compander achieved a very mild gain error recovery due to the natural truncation rounding error caused by twelve bit arithmetic. The complete function of delta modulation, VAC and Echo Control for six ports was implemented in a single digital integrated circuit chip with twelve bit arithmetic. A single digital-to-analog converter (DAC)

600-503: The half second echo path through the satellite. They performed formal listening tests to verify the 24 kbit/s delta modulator achieved full voice quality with no discernible degradation as compared to a high quality phone line or the standard 64 kbit/s μ-law companded PCM. This provided an eight to three improvement in satellite channel capacity. IBM developed the Satellite Communications Controller and

630-431: The integrator with a Predictor implemented with a two pole complex pair low-pass filter designed to approximate the long term average speech spectrum. The theory was that ideally the integrator should be a predictor designed to match the signal spectrum. A nearly perfect Shindler Compander replaced the modified version. It was found the modified compander resulted in a less than perfect step size at most signal levels and

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660-410: The maximum amplitude of the input signal can be A m a x = σ f s ω {\displaystyle A_{max}={\sigma f_{s} \over \omega }} , where f s is the sampling frequency and ω is the frequency of the input signal and σ is step size in quantization. So A max is the maximum amplitude that DM can transmit without causing

690-491: The modulator is | m ˙ ( t ) | m a x = ω A {\displaystyle |{\dot {m}}(t)|_{max}=\omega A} , whereas the condition to avoid slope overload is | m ˙ ( t ) | m a x = ω A < σ f s {\displaystyle |{\dot {m}}(t)|_{max}=\omega A<\sigma f_{s}} . So

720-527: The slope overload and the power of transmitted signal depends on the maximum amplitude. If the communication channel is of limited bandwidth, there is the possibility of interference in either DM or PCM . Hence, 'DM' and 'PCM' operate at same bit-rate which is equal to N times the sampling frequency. The seminal paper combining feedback with oversampling to achieve delta modulation was by F. de Jager of Philips Research Laboratories in 1952. Initial patents include: Adaptive delta modulation (ADM)

750-701: 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 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

780-509: 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

810-403: The transmitted signal has excessive derivative (abrupt changes) then slope overload occurs and the modulated signal can not track the input signal. E.g. if the input signal is m ( t ) = A cos ⁡ ( ω t ) {\displaystyle m(t)={A\cos(\omega t)}} , the modulated signal (derivative of the input signal) which is transmitted by

840-494: The voice port functions. The original proposal in 1974, used a state-of-the-art 24 kbit/s delta modulator with a single integrator and a Shindler Compander modified for gain error recovery. This proved to have less than full phone line speech quality. In 1977, one engineer with two assistants in the IBM Research Triangle Park , NC laboratory was assigned to improve the quality. The final implementation replaced

870-483: 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

900-488: Was first published by Dr. John E. Abate ( Bell Labs Fellow) in his doctoral thesis at NJ Institute Of Technology in 1968. ADM was later selected as the standard for all NASA communications between mission control and space-craft. In the mid-1980s, Massachusetts audio company DBX marketed a commercially unsuccessful digital recording system based on adaptive delta modulation. See DBX 700 . Adaptive delta modulation or Continuously variable slope delta modulation (CVSD)

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