Stephane Villette

Quality Evaluation of Color Plus Depth Map-Based Stereoscopic Video

In the near future, many conventional video applications are likely to be replaced by immersive video to provide a sense of ldquobeing there.rdquo This transition is facilitated by the recent advancement of 3D capture, coding, transmission, and display technologies. Stereoscopic video is the simplest form of 3D video available in the literature. ldquoColor plus depth maprdquo based stereoscopic video has attracted significant attention, as it can reduce storage and bandwidth requirements for the transmission of stereoscopic content over communication channels. However, quality assessment of coded video sequences can currently only be performed reliably using expensive and inconvenient subjective tests. To enable researchers to optimize 3D video systems in a timely fashion, it is essential that reliable objective measures are found. This paper investigates the correlation between subjective and objective evaluation of color plus depth video. The investigation is conducted for different compression ratios, and different video sequences. Transmission over Internet protocol (IP) is also investigated. Subjective tests are performed to determine the image quality and depth perception of a range of differently coded video sequences, with packet loss rates ranging from 0% to 20%. The subjective results are used to determine more accurate objective quality assessment metrics for 3D color plus depth video.

Multiple Description Coding for Voice over IP using Sinusoidal Speech Coding

CELP coders, such as G.729, are often used in VoIP systems as they offer good speech quality in the absence of packet losses. However, their reliance on long-term prediction causes propagation of errors across speech frames, and therefore makes CELP coders more sensitive to packet losses. Sinusoidal coders on the other hand do not rely on long-term prediction, and may be a good alternative for VoIP due to their higher resilience to packet losses. In this paper a comparison is made between CELP and sinusoidal coders in a VoIP application. A packetisation scheme based on multiple description coding (MDC) applied to the sinusoidal coder is presented. The results show that under typical VoIP operating conditions, the sinusoidal coder based systems can outperform CELP based systems at equal bit rate, especially for high packet loss rates

LSF quantisation for pitch synchronous speech coders

The quality of low bit-rate speech coders is reduced at transitions where speech spectral characteristics vary significantly, as usual speech parameter interpolation assumptions fail to correctly model such variations. This paper presents a joint quantisation-interpolation algorithm for coding of LPC parameters in pitch synchronous speech coders to model the rapidly evolving parameters. In this technique a number of sets of pitch synchronous LPC parameters, corresponding to a frame of speech, are jointly coded by coding two reference sets of LSFs and an interpolation trajectory. Coding an interpolation function allows the parameters to vary within the set. The proposed joint quantisation-interpolation coding of the pitch synchronous LSF is evaluated by comparison with time synchronous extraction and linear interpolation. It is also compared with linear interpolation between sets of pitch synchronous LSFs. Comparison results show that the joint quantisation-interpolation method reduces the average spectral distortion when compared to fixed interpolation. The proposed quantiser was included in to the PS-SBLPC coder and informal listening tests carried out. The synthesised speech was found to be of better quality when joint quantisation-interpolation is used.

Split band LPC based adaptive multi-rate GSM candidate

The European Telecommunications Standards Institute (ETSI) has launched a competition for a new mobile communications standard designed to provide better performance than the current GSM standard. This standard is to be called AMR for adaptive multi-rate: the source and channel coding rates can be adapted depending on the state of the channel, thus providing optimal balance between them at any time. The University of Surrey has submitted a candidate for this competition through the Mobile VCE. This candidate was the only one amongst eleven to use a vocoder in the half-rate GSM channel instead of a CELP based coder. The testing which took place as part of the first stage of the competition has shown that this candidate was among the best. This paper presents the system submitted for the half-rate channel as well as the results of the testing.

Efficient parameter quantisation for 2.4/1.2 kb/s split-band LPC coding

Speech coding at very low bit rates has many applications such as answering machines, IP telephony, mobile communications, military communications etc. Most low bit rate coders operate at around 2.4 kb/s, as the speech quality degrades too much below this bit rate. We describe a frequency domain speech coder capable of operating at both 2.9 and 1.2 kb/s, and produces good quality synthesised speech. Both rates use the same analysis and synthesis building blocks over 20 ms, but the 1.2 kb/s coder jointly quantises three sets of parameters every 60 ms to reduce the bit rate while maintaining speech quality. We also describe the quantisation methods used to lower the bit rate from 2.4 kb/s to 1.2 kb/s while retaining most of the quality of the higher bit rate version.

Improved LSF estimation through anti-aliasing filtering

Vocoders compress speech by estimating model parameters at a given transmission rate over an analysis window, assuming that speech is stationary within this window. In this paper, the limits of this assumption are explored with regard to the spectral envelope parameters in the form of line spectral frequency (LSF) parameters. It is shown that all LSF parameters have considerable variations over time, regardless of LSF vector extraction and transmission rates. LSF track variations are investigated through oversampling and are shown to contain high frequency variations above the frequency corresponding to the LSF vector transmission rate. An anti-aliasing filter with cut-off frequency adequate for the chosen LSF vector transmission rate is proposed to alleviate possible spectral overlapping of the LSF parameter spectra. It is confirmed, through experiments, that the proposed method offers an advantage over the classic LSF extraction method with respect to quantisation shown by bit savings of typically 10 to 15%.

Pitch synchronous split-band LPC (PS-SBLPC) vocoder

A pitch-synchronous split band LPC (PS-SBLPC) speech coder is proposed. In this new paradigm, harmonic analysis is carried out on individual pitch cycle waveforms (PCWs) rather than using a large window. PCWs are identified using a trapezoidal window search performed on a modified time envelope signal. In order to achieve a fixed rate coder the PCW parameters are jointly quantised using a combined interpolation and quantisation routine. Combining interpolation and quantisation allows for high correlation between successive PCWs to be exploited, without subjecting rapid transitions to time smoothing. During speech synthesis, no interpolation is applied, as parameter smoothing is provided during the quantisation. Simulation results comparing the PS-SBLPC model with the SB-LPC model show that the quality of the PS-SBLPC speech signal is significantly better than that of the split band LPC (SB-LPC). Initial results have shown that quantisation optimisation leads to vast improvements in speech quality during speech transitions.

Index assignment-based channel coding

A scheme named index assignment-based channel coding (IACC) has been developed for resilience of speech/audio codecs against the bit errors commonly experienced in wireless channels. Although IACC is a type of joint source channel coding, it does not intervene with the source codec design. The proposed scheme takes into account source characteristics and adjusts the amount of coding according to the sensitivity of different values of the source parameters. It is shown that source characteristics play an important role in the performance of IACC. The performance of IACC has been evaluated by applying it to parameters generated by adaptive multi-rate wideband (AMR-WB+) audio codec. A method for perceptual training of IACC codes is also proposed. The results demonstrate that the performance of IACC and IACC concatenated with convolutional coding can be superior to that of conventional convolutional coding at high and moderate bit error rates, respectively.

An Objective Evaluation Methodology for Blind Bandwidth Extension.

In this paper we introduce an objective evaluation methodology for Blind Bandwidth Extension (BBE) algorithms. The methodology combines an objective method, POLQA, with a bandwidth requirement, based on a frequency mask. We compare its results to subjective test data, and show that it gives consistent results across several bandwidth extension algorithms. Additionally, we show that our latest BBE algorithm achieves quality similar to AMR-WB at 8.85 kbps, using both subjective and objective evaluation methods.

Quantisation for Multiple Description Coding for voice over IP

The transmission of voice over IP networks is heavily affected by packet losses. An increasingly popular method to increase the error resilience of these systems is the use of multiple description coding (MDC). However, the MDC techniques commonly used tend to add a significant amount of redundancies, which are not always easy to use optimally. In this paper, we propose a simple vector quantisation scheme to maximise MDC performance, and study several factors affecting its performance under various error conditions. The results show that it is possible to obtain good performance under packet loss conditions, while using only limited amounts of redundancy.