A new noise production and propagation model for open- and closed loop linear predictive coding (LPC) is proposed in this paper. The model allows to accurately predict the overall SNR even at lower bit rates where the conventional high rate theory fails. Moreover, a source of LPC encoder instabilities is pointed out which is due to the interaction between the quantizer and the (filtered) feedback of the quantization error. The new model is verified by measurements.

A new method for wideband speech transmission is proposed which is fully backwards compatible with narrowband telephone systems. For this purpose, a pitch-scaled version of the higher speech frequencies (4 - 6.4 kHz) is inserted into the previously “unused” 3.4 - 4 kHz frequency range of standard telephone speech. This operation is reverted at the decoder side. A consistently good wideband speech quality can be achieved, even after transmission over common codecs and codec tandems. The quality impact on the narrowband part of the signal is insignificant.

This paper presents a recovery scheme for the error-propagation distortion which frequently appears after a frame erasure in CELP-based speech coders, in particular the AMR codec. The extensive use of predictive filters and parameter encoding allow a high-quality speech synthesis in these codecs, but makes them more vulnerable to frame erasures. Thus, when a frame is lost, an additional distortion appears in the subsequent frame, although that was correctly received, further degrading the speech quality. This degradation can also propagate over several frames, being even more damaging than the loss itself. This well known fact has motivated the development of techniques which prevent or mitigate the error propagation. Nevertheless, the previously proposed methods in some respect modify the transmission scheme (by including additional frames, FEC codes, etc.) making them incompatible with the original decoder. In this work, we apply a steganographic technique to embed recovery data to assist the decoder after a frame loss. This data mainly consist of resynchronization pulses and correction vectors for the excitation signal and the spectral envelope, respectively. PESQ results confirm that our proposal achieves a higher robustness against error propagation while the full backwards-compatibility with the AMR standard is retained.

This paper introduces a novel and highly efficient realization of a spherical vector quantizer (SVQ), the "Gosset Low Complexity Vector Quantizer" (GLCVQ). The GLCVQ codebook is composed of vectors that are located on spherical shells of the Gosset lattice E8. A high encoding efficiency is achieved by representing the spherical vector codebook as aggregated permutation codes. Compared to previous algorithms, the computational complexity and memory consumption is further reduced by exploiting the properties of so called classleader root vectors and by a novel approach for the codevector-to-index-mapping. The GLCVQ concept can be generalized to vector dimensions that are multiples of eight. In particular, GLCVQ for 16-dimensional vectors is used in Amd. 6 to ITU-T Rec. G.729.1.

This contribution presents a wideband (50 Hz-7 kHz) speech enhancement system that is operating in the frequency domain. As a novel feature, techniques known from artificial bandwidth extension (BWE) are used to improve the spectral estimation process by exploiting the statistical dependencies between the low band (50Hz - 4kHz) and the high band (4-7kHz). Conventional noise suppression is used in the low band, while a novel approach is applied to the high band. Features from the processed (enhanced) low band signal are extracted and used to estimate subband energies of the high band. The weighting gains determined from these energy estimates are adaptively combined with conventional gains obtained in addition for the high band. The performance of the proposed method is shown to be consistently better than the conventional approach, especially at low input SNR values.

Artificial bandwidth extension (ABWE) of speech signals aims to estimate wideband speech (50 Hz – 7 kHz) from narrowband signals (300 Hz – 3.4 kHz). Applying the source-filter model of speech, many existing algorithms estimate vocal tract filter parameters independently of the source signal. However, many current methods for extending the narrowband voice source signal are limited to straightforward signal processing techniques which are only effective for high-band estimation. This paper presents a method for ABWE that employs novel data-driven modelling and an existing spectral mirroring technique to estimate the wideband source signal in both the high and low extension bands. A state-of-the-art Hidden Markov Model-based estimator evaluates the temporal and spectral envelopes in the missing frequency bands, with which the ABWE speech signal is synthesized. Informal listening tests comparing two existing source estimation techniques and two permutations of the proposed approach show an improvement in the perceived bandwidth of speech signals, in particular towards low frequencies. Subjective tests on the same data show a preference for the proposed techniques over the existing methods under test.

This paper describes the speech and audio codec that has been submitted to ITU-T by Huawei and ETRI as a candidate for the upcoming super-wideband and stereo extensions of Rec. G.729.1 and G.718. The core codec in the current implementation is G.729.1 and the encoded frequency range is increased from 7 kHz to 14 kHz. Therefore, the maximum bit rate is raised from 32 kbit/s to 64 kbit/s by adding five bitstream layers. A comprehensive overview of the codec is presented with a focus on the mono coding components. The results of the listening tests that have been conducted during the ITU-T qualification phase are summarized. The proposed codec passes all quality requirements for mono input signals.

A new method for hiding digital data in the bitstream of an ACELP speech codec is proposed in this paper. The key element of our method is an alternative search strategy for the ACELP codebook which allows for joint data hiding and speech coding. The concept has been examplarily applied to the AMR speech codec (12.2 kbit/s mode) and it is shown that steganographic data can be reliably transmitted at a rate of up to 2 kbit/s both with a negligible effect on the subjective quality of the coded speech and with reasonable computational complexity. Apart from data hiding, it is further pointed out that our method can also be exploited to reduce the codec bit rate.

We consider the problem of transmitting a wideband speech signal with a cut-off frequency of fc= 7 kHz over a standardized narrowband (fc= 3.4 kHz) communication link in a backwards compatible manner. In a previous contribution we have shown that backwards compatibility can be achieved by using digital watermarking: we embedded compact side information about the missing high frequency band (3.4 - 7 kHz) into the narrowband speech signal. Here, we present a related system which is especially tailored to state-of-the-art narrowband speech communication networks such as GSM or UMTS. Therefore, we propose an improved low bit rate bandwidth extension algorithm along with a robust watermarking scheme for CELP-type speech codecs. The practical relevance of our system is shown by speech quality evaluations and by link-level simulations for the "enhanced full rate traffic channel" (TCH/EFS) of the GSM cellular communication system.

This paper describes the scalable coder - G.729.1 - which has been recently standardized by ITU-T for wideband telephony and voice over IP (VoIP) applications. G.729.1 can operate at 12 different bit rates from 32 down to 8 kbit/s with wideband quality starting at 14 kbit/s. This coder is a bitstream interoperable extension of ITU-T G.729 based on three embedded stages: narrowband cascaded CELP coding at 8 and 12 kbit/s, time-domain bandwidth extension (TDBWE) at 14 kbit/s, and split-band MDCT coding with spherical vector quantization (VQ) and pre-echo reduction from 16 to 32 kbit/s. Side information - consisting of signal class, phase, and energy - is transmitted at 12, 14 and 16 kbit/s to improve the resilience and recovery of the decoder in case of frame erasures. The quality, delay, and complexity of G.729.1 are summarized based on ITU-T results.

This paper discusses a potential extension of the ITU-T G.729.1 speech and audio codec. The G.729.1 coder is hierarchically organized, i.e., the obtained quality increases with the amount of bits that is received for each frame. In particular, the bit rates of 8 and 12 kbit/s offer narrowband (50 Hz – 4 kHz) speech transmission. With a received bit rate of at least 14 kbit/s, the output bandwidth is increased to the wideband frequency range (50 Hz – 7 kHz). Here, we investigate efficient methods to provide the full wideband frequency bandwidth already for the lower bit rates of 8 and 12 kbit/s while maintaining interoperability with the standard implementation of G.729.1. These techniques are not necessarily limited to G.729.1 and thus may serve in other applications as well.

Recommendation G.729.1 is a new ITU-T standard which was approved in May 2006. This recommendation describes a hierarchical speech and audio coding algorithm built on top of a narrowband core codec. One challenge in the codec design is the generation of a wideband signal with a very limited additional bit rate (less than 2 kb/s). In this paper, we describe the respective codec layer, which extends the transmitted acoustic bandwidth from the narrowband frequency range (50 Hz-4 kHz) to the wideband frequency range (50 Hz-7 kHz). The underlying algorithm uses a fairly coarse parametric description of the temporal and spectral energy envelopes of the high frequency band (4-7 kHz). This parameter set is quantized with a bit rate of 1.65 kb/s. At the decoder side, the high-frequency components are regenerated by appropriately shaping a synthetically generated ldquoexcitation signal.rdquo Apart from the algorithmic description and a discussion, we state a complexity evaluation as well as some listening test results.

We present a technique to extend narrowband (NB) speech communication systems, using e.g. the GSM enhanced full rate (EFR) codec, with wideband (WB, 50-7000 Hz) capability. The limited acoustic bandwidth of narrowband speech coding is extended using a fairly coarse description of the missing high frequency band (3.4-7 kHz) in terms of temporal and spectral envelopes. The high-band parameters are quantized, transmitted and then used at the receiver side to regenerate the high frequency components. The parameter encoding is done by applying split vector quantization in a transformed domain. This quantization scheme can be scaled to match any given target bit rate. Several example configurations have been implemented and tested in MUSHRA-style listening tests

We present an embedded and hierarchical 8-32 kbit/s speech and audio coding algorithm that has been successfully submitted to the ITU-T as a candidate for ITU-T Rec. G.729.1 (ex G.729EV). The submitting consortium consisted of Siemens AG, Matsushita Electric Industrial Co. Ltd., and Mindspeed Technologies Inc. This contribution gives a comprehensive overview of the proposed codec, describes the implemented algorithms, and states a detailed characterization as well as results of the official G.729EV qualification tests

In recent years research in the area of artificial bandwidth extension (BWE) of speech signals has made significant progress. The intention of BWE is to produce wideband speech with a cut-off frequency of for instance 7 kHz from a narrowband version (e.g., with telephone bandwidth, i.e., 300 Hz − 3.4 kHz). The respective algorithms are based on the estimation of parameters of a source model for speech production given the knowledge of thenarrowband signal. Atheoretical performance bound on this estimation has been formulated in [1]. In order to overcome this boundary we propose the transmission of (compact) side information which can support the parameter estimation. Since a common additional channel would conflict with the requirement of backwards compatibility in narrowband communication systems, the side information is embedded as digital watermark into the narrowband speech.