Subjective Evaluation of Several Efficient Speech Coders

Daumer, W.

doi:10.1109/tcom.1982.1095508

Cited by 54 publications

(9 citation statements)

References 10 publications

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“…Objectives such as delay, video resolution, and audio quality are all affected by the encoding technique used. Figure 4 illustrates approximate tradeoffs between quality (as expressed by a mean opinion score determined via experimentation), required processing resources (correlated with the delay objective), and transmission rates for various audio compression techniques [Daumer, 1982]. As expected, a low-complexity pulse code modulation (PCM) coder must transmit more data than a high-complexity coder to achieve the same subjective quality.…”

Section: Characteristics Of Execution Techniquesmentioning

confidence: 91%

System Resource Management for Distributed Real-Time Systems.

Davis¹,

Klaseen²,

Schreier³

et al. 1995

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Section: Characteristics Of Execution Techniquesmentioning

confidence: 91%

System Resource Management for Distributed Real-Time Systems.

Davis¹,

Klaseen²,

Schreier³

et al. 1995

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“…It is an extremely simple, sample-by-sample nonuniform memoryless quantizer and it achieves what is called toll quality, which is the standard level of performance against which all other narrowband speech coders are judged. As such, a G.711 codec is almost always included in ACR subjective listening tests as a benchmark [14].…”

Section: Waveform Codingmentioning

confidence: 99%

Rate Distortion Bounds for Voice and Video

Gibson

2014

FNT in Communications and Information Theory

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Numerous voice, still image, audio, and video compression standards have been developed over the last 25 years, and significant advances in the state of the art have been achieved. However, in the more than years since Shannon's seminal 1959 paper, no rate distortion bounds for voice and video have been forthcoming. In this volume, we present the first rate distortion bounds for voice and video that actually lower bound the operational rate distortion performance of the best-performing voice and video codecs. The bounds indicate that improvements in rate distortion performance of approximately 50% over the best-performing voice and video codecs are possible. Research directions to improve the new bounds are discussed.

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“…Two male and two female speech waveform samples at 16 kb/s, each for about 3-5 s long, are used in our simulation. The codec performance is evaluated by both an objective measure called the segmental signal-to-noise ratio (SEGSNR) and a subjective measure [22] called the MOS. Different versions of CAPDM are compared with the continuous variable slope delta (CVSD) codec at 16 kb/s [20] and the G.721 ADPCM codec at 32 kb/s.…”

Section: Capdm Performance Evaluation In An Ideal Channelmentioning

confidence: 99%

A packet-based CAPDM speech coder for PCN applications

Liu

Huang²

2000

IEEE Trans. Veh. Technol.

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In this paper, we present a median-rate speech coder, the controlled adaptive prediction delta modulation coder (CAPDM), which operates at 16 kb/s with good speech quality and low algorithm complexity [15]. The coder is dedicated to personal communication network (PCN) applications and transmits speech samples on the basis of packets. It combines the features of one-step looking forward decision, syllabic companding, instantaneous companding, and adaptive prediction. In addition to the use of a short-term prediction filter, CAPDM also exploits the pitch property to predict speech waveform explicitly. With the aid of a pitch prediction filter, the performance of a CAPDM codec improves about 3 dB in segmental signal-to-noise ratio (SEGSNR). The average SEGSNR of CAPDM.FF is about 21 dB, which is 7 dB over traditional CVSD at 16 kb/s. We also utilize an adaptive postfilter (APF) to enhance the perceptual quality of the decoded speech. The mean opinion score (MOS) listening test of CAPDM.FF with APF shows that its average score achieves 4.19, which is as good as G.728 16-kb/s LD-CELP and is comparable with CCITT G.721 32-kb/s ADPCM. The complexity of CAPDM.FF is evaluated to be 8 MIPS, which is much lower than that of LD-CELP and could be further reduced by adopting a smaller correlation window for pitch detection.To solve the problem of packet loss, we developed a packet-based waveform substitution method by reinitializing the codec parameters at the beginning of each packet. The simulation results show that CAPDM.FF could tolerate 5% of packet loss and still keep an SEGSNR at 10 dB and an MOS at about 3.0.

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Subjective Evaluation of Several Efficient Speech Coders

Cited by 54 publications

References 10 publications

System Resource Management for Distributed Real-Time Systems.

System Resource Management for Distributed Real-Time Systems.

Rate Distortion Bounds for Voice and Video

A packet-based CAPDM speech coder for PCN applications

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