Comparison Category Rating Research Articles

Cochlear Implant Compression Optimization for Musical Sound Quality in MED-EL Users.

Variations in loudness are a fundamental component of the music listening experience. Cochlear implant (CI) processing, including amplitude compression, and a degraded auditory system may further degrade these loudness cues and decrease the enjoyment of music listening. This study aimed to identify optimal CI sound processor compression settings to improve music sound quality for CI users. Fourteen adult MED-EL CI recipients participated (Experiment No. 1: n = 17 ears; Experiment No. 2: n = 11 ears) in the study. A software application using a modified comparison category rating (CCR) test method allowed participants to compare and rate the sound quality of various CI compression settings while listening to 25 real-world music clips. The two compression settings studied were (1) Maplaw, which informs audibility and compression of soft level sounds, and (2) automatic gain control (AGC), which applies compression to loud sounds. For each experiment, one compression setting (Maplaw or AGC) was held at the default, while the other was varied according to the values available in the clinical CI programming software. Experiment No. 1 compared Maplaw settings of 500, 1000 (default), and 2000. Experiment No. 2 compared AGC settings of 2.5:1, 3:1 (default), and 3.5:1. In Experiment No. 1, the group preferred a higher Maplaw setting of 2000 over the default Maplaw setting of 1000 (p = 0.003) for music listening. There was no significant difference in music sound quality between the Maplaw setting of 500 and the default setting (p = 0.278). In Experiment No. 2, a main effect of AGC setting was found; however, no significant difference in sound quality ratings for pairwise comparisons were found between the experimental settings and the default setting (2.5:1 versus 3:1 at p = 0.546; 3.5:1 versus 3:1 at p = 0.059). CI users reported improvements in music sound quality with higher than default Maplaw or AGC settings. Thus, participants preferred slightly higher compression for music listening, with results having clinical implications for improving music perception in CI users.

Subjective and objective evaluation of sound quality of radio programs transmitted via Digital Audio Broadcast (DAB+) System

The work presents results of research on the sound quality of different radio programs transmitted via Digital Audio Broadcasting (DAB+). This assessment has been provided with a use of psychoacoustic model as well as standard listening tests, using an Absolute Category Rating (ACR) method of scaling, and Comparison Category Rating (CCR) method. Results have shown that sound quality gets worse when bit-stream is of the lowest values (48 kbit/s or 24 kbit/s). Application of the Spectral Band Replication processor significantly improves the perceived quality, which is satisfying for bit streams higher than 64 kbit/s, particularly for jazz and popular music. The assessment with ACR method (recommended for broadcast by International Telecommunication Union) showed better notes than CCR one. It means that recommended method is less critical. Also results obtained with psychoacoustic model are more similar to obtained with CCR method. The attributes of spatial impression change in different ways. The greatest distortion has been observed for the perspective and spaciousness of sound image, while the sound color as well as localization stability and accuracy of phantom sources remained almost the same.

Frame Splitting Scheme for Error-Robust Audio Streaming over Packet-Switching Networks

This letter presents a novel frame splitting scheme for an error-robust audio streaming over packet-switching networks. In our approach to perceptual audio coding, an audio frame is split into several subframes based on the network configuration such that each packet can be decoded independently at the receiver. Through a subjective comparison category rating (CCR) test, it is discovered that our approach enhances the quality of the decoded audio signal under the lossy packet-switching networks environment.

Understanding perceptual distortion in MPEG scalable audio coding

In this paper, we study coding artifacts in MPEG-compressed scalable audio. Specifically, we consider the MPEG advanced audio coder (AAC) using bit slice scalable arithmetic coding (BSAC) as implemented in the MPEG-4 reference software. First we perform human subjective testing using the comparison category rating (CCR) approach, quantitatively comparing the performance of scalable BSAC with the nonscaled TwinVQ and AAC algorithms. This testing indicates that scalable BSAC performs very poorly relative to TwinVQ at the lowest bitrate considered (16 kb/s) largely because of an annoying and seemingly random mid-range tonal signal that is superimposed onto the desired output. In order to better understand and quantify the distortion introduced into compressed audio at low bit rates, we apply two analysis techniques: Reng bifrequency probing and time-frequency decomposition. Using Reng probing, we conclude that aliasing is most likely not the cause of the annoying tonal signal; instead, time-frequency or spectrogram analysis indicates that its cause is most likely suboptimal bit allocation. Finally, we describe the energy equalization quality metric (EEQM) for predicting the relative perceptual performance of the different coding algorithms and compare its predictive ability with that of ITU Recommendation ITU-R BS.1387-1.

A very low bit rate speech coder based on a recognition/synthesis paradigm

Previous studies have shown that a concatenative speech synthesis system with a large database produces more natural sounding speech. We apply this paradigm to the design of improved very low bit rate speech coders (sub 1000 b/s). The proposed speech coder consists of unit selection, prosody coding, prosody modification and waveform concatenation. The encoder selects the best unit sequence from a large database and compresses the prosody information. The transmitted parameters include unit indices and the prosody information. To increase naturalness as well as intelligibility, two costs are considered in the unit selection process: an acoustic target cost and a concatenation cost. A rate-distortion-based piecewise linear approximation is proposed to compress the pitch contour. The decoder concatenates the set of units, and then synthesizes the resultant sequence of speech frames using the harmonic+noise model (HNM) scheme. Before concatenating units, prosody modification which includes pitch shifting and gain modification is applied to match those of the input speech. With single speaker stimuli, a comparison category rating (CCR) test shows that the performance of the proposed coder is close to that of the 2400-b/s MELP coder at an average bit rate of about 800-b/s during talk spurts.