Showing papers on "Enhanced Variable Rate Codec published in 2015"

Overview of the EVS codec architecture

Abstract: 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.

Standardization of the new 3GPP EVS codec

Abstract: 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.

Packet-loss concealment technology advances in EVS

Abstract: EVS, the newly standardized 3GPP Codec for Enhanced Voice Services (EVS) was developed for mobile services such as VoLTE, where error resilience is highly essential. The presented paper outlines all aspects of the advances brought during the EVS development on packet loss concealment, by presenting a high level description of all technical features present in the final standardized codec. Coupled with jitter buffer management, the EVS codec provides robustness against late or lost packets. The advantages of the new EVS codec over reference codecs are further discussed based on listening test results.

Subjective quality evaluation of the 3GPP EVS codec

Abstract: This paper discusses the voice and audio quality characteristics of EVS, the recently standardized 3GPP codec. Comparison to Opus, IETF driven open source codec as well as industry standard voice codecs: 3GPP AMR and AMR-WB, and ITU-T G.718B, G.722.1C and G.719 as well as direct signals at varying bandwidths was made. Voice and audio quality was evaluated with three subjective listening tests containing clean and noisy speech in Finnish language as well as a mixed condition test containing both speech and music intermixed. Nine-scale subjective mean opinion score was calculated for all tested conditions.

Super-wideband bandwidth extension for speech in the 3GPP EVS codec

Abstract: 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.

Speech coding and enhancement using quantized compressive sensing measurements

Abstract: Medium bit rate hybrid speech coding schemes have gained much interest in the recent years and many of them have been standardized for various applications. This work characterizes a speech codec in a Compressive Sensing (CS) framework. We mainly demonstrate two aspects 1) Simultaneous compression and de-noising of speech by CS 2) Appropriate quantization of CS measurements to design medium bit-rate codec. The proposed scheme renders better quality speech compared to CELP, the widely used hybrid coding scheme, at the same bit rates. The CS speech codec has the added advantage of inherent noise suppression and easy scalability, without complex parameter extractions and voice activity detections.

System aspects of the 3GPP evolution towards enhanced voice services

Abstract: The Enhanced Voice Services (EVS) codec was standardized by 3GPP in 2014. This codec offers significant gains in voice quality, efficiency, channel error robustness over any other existing speech codec and far better music quality. Operators run voice services on a large installed base of 3GPP Circuit-Switched (CS) 2G (GERAN) or 3G (UTRAN) radio networks. These networks offer mobile voice service either as HD voice using the AMR-WB codec, or as traditional narrowband (NB) voice service, based on the AMR codec. Voice over LTE (VoLTE) is currently being deployed throughout the world with HD Voice. The EVS codec will be first introduced as a straightforward VoLTE upgrade. It is also expected that EVS will be deployed over 3G CS networks. This paper describes system aspects relating to EVS codec introduction in VoLTE and CS networks as well as interworking and mobility with legacy systems and services.

Impact of the codec and various QoS methods on the final quality of the transferred voice in an IP network

Abstract: This paper deals with an analysis of the relation between the codec that is used, the QoS method, and the final voice transmission quality. The Cisco 2811 router is used for adjusting QoS. VoIP client Linphone is used for adjusting the codec. The criterion for transmission quality is the MOS parameter investigated with the ITU-T P.862 PESQ and P.863 POLQA algorithms.

Evaluation of narrow band speech codecs for Ubiquitous Speech Collection and Analysis Systems

Abstract: With wireless acoustic sensor network extending to the services like surveillance of sensitive areas, such as Line of Control, or unmanned terrains, interest in robust, narrowband and low bit rate speech codecs is increasing. This has resulted in a need for evaluation of such codecs. This work investigates the different factors like bit rate, algorithmic delay, implementation, and more importantly robustness of codec's perceptual quality to noise. These factors play a vital role in choosing a speech codec and evaluating its performance. This work examines the selected narrow band speech codecs using ITU-T (International Telecommunication Union) recommendation P.862 tool. This study can also help to choose a speech codec in myriad applications like Wireless Acoustic Sensor Networks, Cellular Telephony, VoIP, Underwater Acoustic Monitoring and many other applications where bandwidth plays a vital role and data compression is inevitable.

The Optimization of G.729 Speech codec and Implementation on the TMS320VC5402

Abstract: This paper designs a realtime speech codec system based on the TMS320C5402 platform, from two aspects of data transmission interface and control interface for the TMS320C5402 and TLV320AIC23 interfacedesign, and through the g. 729 speech decoding algorithm principle research and program optimization on the platform to realize the real-time algorithm.

Audio bandwidth detection in the EVS codec

Abstract: 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.

A packet loss recovery technique with line spectral frequency modification in 3GPP EVS codec

Abstract: This paper examines a method for controlling the energy of decoded signal at the recovery frame from a packet loss. Our observation unveiled that a packet loss before speech onset causes sudden increase in the amplitude of the decoded signal at the recovery frame when predictive quantization of line spectral frequency is used. To mitigate the artifact caused by the overshoot, a detector of the overshoot is proposed as well as a method that controls the amplitude of the decoded signal by adjusting distances of adjacent line spectral frequencies. This technology is implemented in the enhanced voice services (EVS) codec which is the latest 3GPP speech and audio codec standard.