Pasi Ojala

AMR-WB+: a new audio coding standard for 3rd generation mobile audio services

Highly efficient low-rate audio coding methods are required for new compelling and commercially interesting applications of streaming, messaging and broadcasting services using audio media in 3rd generation mobile communication systems. After an audio codec selection phase, 3GPP has standardized the extended AMR-WB (AMR-WB+) codec that provides a unique performance at very low bit rates from below 10 kbps up to 24 kbps. This paper discusses the requirements imposed by mobile audio services and gives a technology overview of AMR-WB+ as a codec matching these requirements while providing outstanding audio quality.

Media coding for the next generation mobile system LTE

Introduction of LTE (Long Term Evolution) brings enhanced quality for 3GPP multimedia services. The high throughput and low latency of LTE enable higher quality media coding than what is possible in UMTS. LTE-specific codecs have not yet been defined but work on them is ongoing in 3GPP. The LTE codecs are expected to improve the basic signal quality, but also to offer new capabilities such as extended audio bandwidth, stereo and multi-channels for voice and higher temporal and spatial resolutions for video. Due to the wide range of functionalities in media coding, LTE gives more flexibility for service provision to cope with heterogeneous terminal capabilities and transmission over heterogeneous network conditions. By adjusting the bit-rate, the computational complexity, and the spatial and temporal resolution of audio and video, transport and rendering can be optimised throughout the media path hence guaranteeing the best possible quality of service.

The effect of source based rate adaptation extension in AMR-WB speech codec

This paper presents a source based rate adaptation concept for AMR wideband speech codec. The source based rate adaptation algorithm selects the multi rate codec mode based on the input speech characteristics and coding parameters to minimise the average bit rate. The presented concept introduces up to 50% reduction in average bit rate without any degradation in speech quality. The benefit of source based adaptation is in increasing the system capacity in conversational services as well as storage size in messaging type of applications.

The adaptive multirate wideband speech codec: system characteristics, quality advances, and deployment strategies

Wideband speech is the major differentiation and attraction of third-generation network services in both the circuit and packet switched domain. Increased audio bandwidth introduces a significant leap in perceived quality of service compared to currently utilized narrowband telephony in second-generation mobile communications and the PSTN. The adaptive multirate wideband (AMR-WB) speech codec is the service enabler for improved user experience. It is an established 3GPP and ITU-T wideband speech codec standard and represents the state-of-the-art in speech quality as well as robustness in error prone radio channels. It is also the first codec algorithm standardized for wideband speech for mobile communications

Toll quality variable-rate speech codec

This paper presents a source controlled variable-rate CELP type speech codec. First, a voice activity detection block distinguishes active speech frames from silence and background noise. The active speech is further classified into voiced and unvoiced frames. The voiced frames have variable bit-rate pitch-lag quantization based on the characteristics of the speech, whereas the unvoiced frames are coded without pitch information. A variable bit-rate fixed codebook excitation with a variable number of excitation pulses is determined for each speech frame. The performance of the linear analysis part of the codec as well as the input speech characteristics determine the excitation bit-rate. The average bit-rate of the codec is around 7.0 kbit/s for active speech, and the overall bit-rate ranges from 0 to 7.85 kbit/s. The described variable-rate codec produces toll quality speech equal to that of the 32 kbit/s ADPCM (G.726) standard.

A novel pitch-lag search method using adaptive weighting and median filtering

This paper presents a novel method to estimate the pitch-lag in a speech codec. The pitch-lag is related to the fundamental frequency of the speech signal and an accurate estimation of this parameter is important for the subjective quality of the synthesised speech. A common problem in speech codecs is that the estimation of the pitch-lag often produces a multiple or a sub-multiple of the true pitch value. When these incorrect pitch-lag values are used in speech synthesis the subjective quality of the speech is degraded. This paper presents an improved method where the estimation of the pitch-lag parameter is biased towards the pitch-lag values of the previous speech segments resulting in a consistent set of consecutive pitch-lag values and a high quality reconstructed signal. The classification of speech into voiced and unvoiced parts is used when tracking the pitch-lag values and adapting the pitch track centered weighting function.

Variable model order LPC quantization

This paper presents a new method to apply variable bit-rate predictive quantization of the variable model order LPC parameters. In addition, the method is employed to interpolate the parameters within the analysis frame. The LPC model order selection algorithm is based on the characteristics of the input signal and on the performance of the LPC model. Hence, the variable bit-rate LPC quantization is source controlled. The number of quantized parameters needs to be identical in successive frames to be able to apply the predictive quantization and to interpolate parameters inside the frame. Therefore, the order of the LPC model of the previous frame needs to be expanded or reduced to be the same as the current frame LPC model. The advantage of variable model order LPC quantization is the lowered average bit-rate compared to a fixed rate while the speech quality remains the same.

Low Complex Audio Encoding for Mobile, Multimedia

Computational resources in a mobile terminal and in wireless transmission channels set strict requirements for audio coding in mobile multimedia applications. This paper introduces, a low complex audio encoding method enabling high-quality audio for mobile multimedia. The presented method is the low complexity version of the extended AMR-WB audio encoder in the 3GPP standards. The encoder was designed based on requirements of the 3GPP multimedia messaging service (MMS)

Parametric binaural audio coding

A spatial audio scene consists of discrete audio sources and ambience. The 3D audio image is observed due to the directional sounds, but even more important is the reverberation and so called room effect caused by the properties of the space, the sources and listener location. It is obvious that a human being is able to capture the 3D image using the signal from left and right ear. Hence, two audio channels are sufficient to represent the spatial audio image for the listener. An efficient transmission and representation of a spatial audio image using two channels requires a specific coding and rendering algorithm for the audio content. In this paper we present novel mechanisms to efficiently parameterize, quantize and represent the spatial audio signal.

Assessing Value of SW Requirements

Understanding software requirements and customer needs is vital for all SW companies around the world. Lately clearly more attention has been focused also on the costs, cost-effectiveness, productivity and value of software development and products. This study outlines concepts, principles and process of implementing a value assessment for SW requirements. The main purpose of this study is to collect experiences whether the value assessment for product requirements is useful for companies, works in practice, and what are the strengths and weaknesses of using it. This is done by implementing value assessment in a case company step by step to see which phases possibly work and which phases possibly do not work. The practical industrial case shows that proposed value assessment for product requirements is useful and supports companies trying to find value in their products.