Vaclav Eksler

Overview of the EVS codec architecture

The recently standardized 3GPP codec for Enhanced Voice Services (EVS) offers new features and improvements for low-delay real-time communication systems. Based on a novel, switched low-delay speech/audio codec, the EVS codec contains various tools for better compression efficiency and higher quality for clean/noisy speech, mixed content and music, including support for wideband, super-wideband and full-band content. The EVS codec operates in a broad range of bitrates, is highly robust against packet loss and provides an AMR-WB interoperable mode for compatibility with existing systems. This paper gives an overview of the underlying architecture as well as the novel technologies in the EVS codec and presents listening test results showing the performance of the new codec in terms of compression and speech/audio quality.

Classification-Based Techniques for Improving the Robustness of CELP Coders

This paper describes techniques which can be used at the encoder and at the decoder of CELP coders to improve their robustness to frame erasures. The techniques address specific problems in different speech classes, in particular stationary voiced and voiced onsets.

Glottal-Shape Codebook to Improve Robustness of CELP Codecs

This paper presents a new technique for the class of code-excited linear prediction speech codecs designed to reduce error propagation after lost frames. Its principle consists in replacing the interframe long-term prediction with a glottal-shape codebook in the subframe containing the first glottal impulse in a given frame. This technique, independent of previous frames, is of particular interest in voiced speech frames following transitions as these frames are the most sensitive to frame erasures. It is a basis of a structured coding scheme called transition coding (TC). The TC greatly improves codec performance in noisy channels while maintaining clean channel performance. It is a part of the new embedded speech and audio codec recently standardized as Recommendation G.718 by ITU-T.

Transition mode coding for source controlled celp codecs

CELP-based codecs typically rely on prediction to achieve their high coding efficiency. On the other hand, the prediction makes these codecs sensitive to frame erasures as errors propagate beyond the erased frame. We present a technique that significantly limits the error propagation by replacing inter-frame long-term prediction with a non-predictive glottal-shape codebook. The technique was implemented in the winning candidate of the EV-VBR baseline codec selection by ITU-T in March 2007. To maintain the performance in clean channel, this transition mode coding technique was used only in frames following voiced onsets frames, i.e. the frames most sensitive to frame errors.

Efficient handling of mode switching and speech transitions in the EVS codec

The recently standardized codec for Enhanced Voice Services (EVS) consists of a number of modes to achieve its high coding flexibility. In this paper we focus on techniques that enable a seamless switching between two linear prediction based modes running at different sampling rates within this codec. The first one deals with an efficient conversion of the linear prediction filter coefficients. The other one is based on a constrained-memory ACELP called transition coding (TC) that significantly limits the inter-frame long-term dependency. We show that the use of TC can be successfully extended to improve quality also in coding other transitions, e.g. strong onsets of voiced speech.

Enhanced AMR-WB bandwidth extension in 3GPP EVS codec

This paper presents the bandwidth extension (BWE) method developed for the AMR-WB interoperable (AMR-WB IO) modes of the 3GPP EVS codec. The low-band signal (0-6.4 kHz) is coded using an enhanced version of ACELP as in AMR-WB and post-processed; the high-band (above 6.4 kHz) in contrast to AMR-WB is represented with a new BWE method. The decoded low-band excitation is adaptively extended to high frequencies and filtered in the DCT domain. The extended excitation is scaled by subframe gains and shaped by a weighted LPC synthesis filter. Test results show that the AMR-WB IO BWE contributes to the performance advantage of EVS AMR-WB IO over the original AMR-WB, with limited complexity increase and no extra delay.

A novel frequency domain BWE with relaxed synchronization and associated BWE switching

This paper presents a novel frequency domain bandwidth extension (BWE) scheme with relaxed synchronization, optimized for coding inactive and music/mixed content signals. The algorithm achieves high subjective quality at low and medium bitrates and it has a low algorithmic delay. The algorithm is part of the 3GPP Enhanced Voice Services (EVS) codec. In addition to the presented algorithm, the EVS codec employs also a time domain BWE scheme optimized for active speech coding. Consequently a seamless switching between these BWE technologies is required and described in this paper as well.

Flexible and scalable transform-domain codebook for high bit rate CELP coders

The Code-Excited Linear Prediction (CELP) model is very efficient in coding speech at low bit rates. However, if the bit rate of the coder is increased, the CELP model does not gain in quality as quickly as other approaches. Moreover, the computational complexity of the CELP model generally increases significantly at higher bit rates. In this paper we focus on a technique that aims to overcome these limitations by means of a special transform-domain codebook within the CELP model. We show by the example of the AMR-WB codec that the CELP model with the new flexible and scalable codebook improves the quality at high bit rates at no additional complexity cost.

Audio bandwidth detection in the EVS codec

Speech and audio codecs are usually designed such that they encode all the frequency bands of the input signal spectrum. If the higher bands do not contain any perceptually meaningful content, these codecs often do not work optimally as they assign part of the available bit budget to encode these bands. In this paper we describe a bandwidth detection algorithm that determines the effective audio bandwidth of the input signal. This information is used to set the codec to its optimal configuration and consequently increase the coding efficiency for band-limited signals by allocating bits to encode only the useful bandwidth. The presented algorithm has been used in the new codec for Enhanced Voice Services (EVS), recently standardized by 3GPP, but it can be employed in other codecs as well.