This paper describes the time-domain bandwidth extension (TBE) framework employed to code wideband and super-wideband speech in the newly standardized 3GPP EVS codec. The TBE algorithm uses a nonlinear harmonic modeling technique that incorporates principles of time-domain envelope-modulated noise mixing. At 13.2 kbps, the super-wideband coding of speech uses as low as 1.55 kbps for encoding the spectral content from 6.4-14.4 kHz. Subjective evaluation results from ITU-T P.800 Mean Opinion Score (MOS) tests are provided, showing significantly improved quality compared to the other standardized SWB codecs under both clean speech and speech with background noise.

A new codec for Enhanced Voice Services (EVS), the successor of the current mobile HD voice codec AMR-WB, was standardized by the 3rd Generation Partnership Project (3GPP) in September 2014. The EVS codec addresses 3GPP's needs for cutting-edge technology enabling operation of 3GPP mobile communication systems in the most competitive means in terms of communication quality and efficiency. This paper provides an in-depth insight into 3GPP's rigorous and transparent processes that made it possible for the mobile industry, with its many competing players, to successfully develop and standardize a codec in an open, fair and constructive process. This paper also enables an understanding of this achievement by providing an overview of the EVS codec technology, the standard specifications, and the performance of the codec that will elevate HD voice services to the next quality level.

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.

Audio coding of harmonic signals is a challenging task for conventional MDCT coding schemes. In this paper we introduce a novel algorithm for improved transform coding of harmonic audio. The algorithm does not deploy the conventional scheme of splitting the input signal into a spectrum envelope and a residual, but models the spectral peak regions. The presented coding scheme is part of the recently standardized 3GPP EVS codec.

This paper describes a novel audio coding algorithm that is a building block in the recently standardized 3GPP EVS codec [1]. The presented scheme operates in the Modified Discrete Cosine Transform (MDCT) domain and deploys a Split-PVQ pulse coding quantizer, a noise-fill, and a gain control optimized for the quantizer's properties. A complexity analysis in terms of WMOPS is presented to illustrate that the proposed Split-PVQ concept and dynamic range optimized MPVQ-indexing are suitable for real-time audio coding. Test results from formal MOS subjective evaluations and objective performance figures are presented to illustrate the competitiveness of the proposed algorithm.

We introduce multivariate block polar quantization (MBPQ). MBPQ minimizes the weighted squared-error distortion for a set of complex variables representing one block of a signal under a resolution constraint for the entire block. MBPQ performs below the lower bound for classical bivariate quantization, both for Gaussian complex variables and for complex variables found from sinusoidal analysis of audio data. Still, it is of similar complexity as traditional polar quantizers. In the case of audio data, we found a performance gain of about 2.5 dB over the best performing conventional resolution-constrained polar quantization (an extension of unrestricted polar quantization).

We introduce multi-variate block polar quantization (MBPQ). MBPQ minimizes a weighted distortion for a set of complex variables representing one block of a signal under a resolution constraint for the entire block. MBPQ allows for different probability distributions in different dimensions of the set of complex variables. It outperforms a block polar quantizer introduced earlier (Pobloth, H. et al., Proc. Eurospeech, p.1097-100, 2003), and unrestricted polar quantization (UPQ), for both Gaussian complex variables and sinusoids found from audio data. In the case of audio data, we found a performance gain of about 2.5 dB over the best performing conventional resolution-constrained polar quantization, UPQ.

In this article, we investigate how accurately the squared error captures perceptual errors introduced by Fourier phase spectrum changes. We measure the perceptual error using the Auditory Image Model by Patterson et al.. The squared error is found to represent the perceptual error well for low squared errors but it saturates. Thus, a further increase in squared error does on average not lead to any further increase in perceptual error. This suggests that encoding phase using squared-error trained codebooks only improves perceived quality when operating at high bit rates. To verify this, phase was encoded with codebooks of different sizes. As expected, increasing the codebook size has very little influence on the average perceptual error for low rates, which is confirmed by listening tests. Our results suggest that a direct phase codebook is an inefficient representation of the relevant information contained in phase.

In this paper we define perceptual phase capacity as the size of a codebook of phase spectra necessary to represent all possible phase spectra in a perceptually accurate manner. We determine the perceptual phase capacity for voiced speech. To this purpose, we use an auditory model which indicates if phase spectrum changes are audible or not. The correct performance of the model was adjusted and verified by listening tests. The perceptual phase capacity in low pitched speech is found to be much higher than it is for high pitched speech. Our results are consistent with the well known fact that speech coding schemes which preserve the phase accurately work better for male voices, while coders which put more weight on the amplitude spectrum of the speech signal result in better quality for female speech.