The ITU-T Recommendation G.711 is the benchmark standard for narrowband telephony. It has been successful for many decades because of its proven voice quality, ubiquity and utility. A new ITU-T recommendation, denoted G.711.0, has been recently established defining a lossless compression for G.711 packet payloads typically found in IP networks. This paper presents a brief overview of technologies employed within the G.711.0 standard and summarizes the compression and complexity results. It is shown that G.711.0 provides greater than 50% average compression in typical service provider environments while keeping low computational complexity for the encoder/decoder pair (1.0 WMOPS average, <;1.7 WMOPS worst case) and low memory footprint (about 5k octets RAM, 5.7k octets ROM, and 3.6k program memory measured in number of basic operators).

This paper proposes a new structure for a scalable codec. Our proposed codec works with 10 ms input frame for wideband speech and audio signals at bit rates ranging from 8 to 32 kbit/s. The core layer is the ITU-T G.729 at 8 kbit/s producing a narrowband output. The first enhancement layer is a band-width extension providing a wideband output with 2 kbit/s. The second enhancement layer is based on algebraic quantization of wavelet packet coefficients and improves gradually the synthesized signal as the bitrate increases. For speech signals, at bitrates of 24 and 32 kbit/s, the codec is shown to be equivalent to the ITU-T G.722 codec at 56 and 64 kbit/s, respectively. Moreover, the codec at 32 kbit/s is assessed to be equivalent to the recently standardized embedded codec ITU-T G.729.1 at the same bitrate with a lower algorithmic delay.

This paper proposes a new quantization for transform coefficients based on algebraic quantization. The coefficients are represented by a few pulses multiplied by a unique amplitude. The coefficients to be transmitted are selected by optimizing an error criterion, that determines the signs, positions and amplitudes of the pulses. This simple quantization has been implemented in a wavelet-based wideband scalable coder, and has been proved to provide a perceptually better quality than SPIHT on speech signal and music.

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

This paper presents a scalable wideband speech codec working at bitrates ranging from 8 to 32 kbit/s. The core layer is the ITU-T G. 729 at 8 kbit/s. A first enhancement layer is a bandwidth extension algorithm requiring 2 kbit/s to widen the G. 729 narrow band output speech. The difference between the wideband original and reconstructed signal is transformed in the time-frequency domain by a full wavelet decomposition. The resulting coefficients are quantized by an embedded quantizer at 22 kbit/s. Listening tests show the relevance of such a scheme when compared to a pure wavelet packet decomposition. In addition, listening tests suggest that the proposed codec is equivalent to the ITU-T G. 722 at 48 kbit/s for speech signals.

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

This paper presents an enhancement layer coming on top of G.729 Annex A codec. This enhancement layer consists of an optimization of the algebraic fixed codebook pulses that were selected by the Annex A codec during the fixed codebook search. The pulses selected by the ACELP codec have all the same absolute amplitude. Our method optimizes each of these amplitudes in order to increase the quality. This is a low-complexity method that allows the introduction of an intermediate bitrate in ACELP-based scalable coder. On top of this pulse optimization method that requires 1.6 kbit/s we introduce a second fixed codebook search with a limited number of pulses (2) and a bitrate of 2.4 kbit/s

This paper examines the application of UDP-lite for unequal error detection in packet-switched speech transmission via Internet protocols (voice-over-IP) over UMTS radio channels. Traditionally, UDP is used as transport layer protocol, which contains a checksum that covers the complete packet. Thus, any packet with residual bit errors is discarded. Speech codecs like AMR, however, can tolerate bit errors in less sensitive parts of the bitstream. A more recent development, UDP-lite, provides unequal error detection with a partial checksum that covers only the sensitive parts of a packet. Thus, only packets with errors in important bits are discarded. We compare the use of UDP-lite for UMTS channels with convolutional and channels with turbo coding. The results show that the achievable quality improvement by applying UDP-lite depends on the residual bit error distribution of the chosen UMTS channel coding method. While we determined a quality improvement for channels with convolutional coding, we did not get an improvement for turbo coding. Furthermore, when combined with header compression, the convolutional coder with use of UDP-lite can reach the performance of the turbo coder with use of UDP

The transmission of speech in mobile or packet networks requires the use of a speech codec. In order to improve the quality of speech in a noisy environment, a noise reduction algorithm is used. This noise reduction can either be done as pre-processing before speech encoding or in the network by decoding the bitstream, performing the speech enhancement in the time and/or frequency domain and re-encoding the speech. Both methods are computationally expensive. In this paper a new approach to reduce environmental background noise by modifying the codec parameters is discussed.

The introduction of packet switched (PS) networks (e.g., IMS) creates a need to evaluate speech transmission quality when using the 3GPP default speech codecs. 3GPP SA4 has created a work item in 3GPP Release 6 on "Performance characterization of default codecs for PS conversational application". France Telecom R&D and Siemens proposed a test framework consisting of a UMTS simulator for the air interface and an IP network simulator. ITU-T recommendation P.800 (1996) is used for the quality estimation of the transmission. The real-time conversational test results show that the AMR-NB (adaptive multirate narrow band) and AMR-WB (AMR wide band) speech codecs are well suited for PS conversational applications. Furthermore, the results clearly show a higher understanding when using AMR-WB rather than AMR-NB.

In this paper we present a voicing controlled, speech parameter based frame loss concealment for frames that have been encoded with the Adaptive Multi-Rate (AMR) speech codec. The missing parameters are estimated by interpolation and extrapolation techniques that are chosen in dependence of the voicing state of the speech frames preceding and following the lost frames. The voicing controlled concealment outperforms the conventional extrapolation/muting based approach and it shows a consistent improvement over interpolation techniques that do not distinguish between voiced and unvoiced speech. The quality can be further improved if additional information about the predictor states of predictively encoded parameters is available from a redundant transmission in future packets.

In this paper we investigate the application of Unequal Error Protection (UEP) channel coding schemes to the Transform Predictive Coding (TPC) audio coding paradigm. The TPC paradigm has natural perceptual ordering of the bitstream that provides a good match with the UEP concept. Also introduced is the general idea of Mode Adaptive UEP schemes that adapt to the differing bit-error sensitivity profiles of signal adaptive multimode source-coders. By doing this the decoded signal quality is optimized jointly over the input signal, the channel conditions and the bit-error sensitivity profile of the source coder.