Showing papers on "Enhanced Variable Rate Codec published in 2013"

Device-initiated codec rate change during a voice call

Abstract: A method for initiating a codec rate change during a VoIP call by a wireless communication device is disclosed. The method can include the wireless communication device establishing a first codec rate for use in the VoIP call during a call establishment phase; using the first codec rate to encode voice data for transmission during a first portion of the VoIP call; determining a channel quality while using the first codec rate; determining that the channel quality satisfies a threshold for requesting a codec rate change; requesting a codec rate change from the first codec rate to a second codec rate in response to the channel quality satisfying the threshold; and using the second codec rate to encode voice data for transmission during a second portion of the VoIP call.

Scalable Speech Coding for IP Networks: Beyond iLBC

Abstract: High quality speech at low bit rates makes code excited linear prediction (CELP) the dominant choice for a narrowband coding technique despite the susceptibility to packet loss. One of the few techniques which received attention after the introduction of CELP coding technique is the internet low bitrate codec (iLBC) because of inherent high robustness to packet loss. Addition of rate flexibility and scalability makes the iLBC an attractive choice for voice communication over IP networks. In this paper, performance improvement schemes of multi-rate iLBC and its scalable structure are proposed, and the proposed codec enhanced from the previous work is re-designed based on the subjective listening quality instead of the objective quality. In particular, perceptual weighting and the modified discrete cosine transform (MDCT) with short overlap in weighted signal domain are employed along with the improved packet loss concealment (PLC) algorithm. The subjective evaluation results show that the speech quality of the proposed codec is equivalent to that of state-of-the-art codec, G.718, under both a clean channel condition and lossy channel conditions. This result is significant considering that development of the proposed codec is still in early stage.

A generic algorithm for mid-call audio codec switching

Abstract: We present and evaluate an algorithm that performs in-call selection of the most appropriate audio codec given prevailing conditions on the network path between the end-points of a voice call We have studied the behaviour of different codecs under varying network conditions, in doing so deriving the impairment factors for non-ITU-T codecs so that the E-model can be used to assess voice call quality for them Moreover, we have studied the drawbacks of codec switching from the end user perception point of view; our switching algorithm seeks to minimise this impact We have tested our algorithm on different packages that contain a selection of the most commonly used codecs: G711, SILK, ILBC, GSM and SPEEX Our results show that in many typical network scenarios, our switching codecs mid-call algorithm results in better Quality of Experience (QoE) than would have been achieved had the initial codec been used throughout the call

Simulasi global system for mobile communication adaptive multi rate speech codec

Abstract: As the wireless technology continuing developing enormously, within it the development of speech signal compression joining too, which is implemented in digital communication system. Among the aim is privileged to minimize the channel capacity or bandwidth usage and the speed of transmission. So the use of smaller bits, which brings maximised voice quality, are the most important factor to this technology. One of the speech signal compression techniques which is implemented in GSM network is called Adaptive Multi Rate (AMR) speech codec or known as GSM-AMR speech codec. There are 8 (eight) bit rates 12.2, 10.2, 7.95, 7.40, 6.70, 5.90, 5.15 and 4.75 kbps which is used by GSM-AMR. The bit rates have abilities to adapt to the situation and condition of the network and also ability to tailor the operator needs and requirement. The result of a non real-time simulation is shown that GSM-AMR has a good enough quality of speech signal reconstruction with range of SNR between 18.78 and 20.35 dB and also range of MSE between 0.053 and 0.079. The smaller the bitrate, the better the SNR and the MSE. It was also proven with the comparison of original signal and constructed signal which is shown that the reconstructed signal has similarities figure with the original signal. Overall, the performance of the speech codec gives a good speech quality. Keyword: Speech Codec, GSM-AMR, Adaptive Multi Rate, SNR, MSE.

Quality codec interface for VoIP on differentiated services networks

Abstract: As converged services are becoming the standard for corporate networks, concerns have been arising on the best way to ensure efficient use of expensive WAN bandwidth. Achieving excellent quality of multimedia and voice communication requires the allocation of large bandwidth for these services. In most cases, lossy compression codecs are used to reduce bandwidth requirements resulting in lower quality of service. The codec selection is maintained even during low bandwidth utilization. In this paper, we present a codec interface that combines high-fidelity high-bitrate and low-fidelity low-bitrate codecs. Audio packets from the low-bitrate codec are transmitted on the highest priority class of service of differentiated services networks. This ensures that these packets always arrive at destination. Audio packets from the high-bitrate codec are transmitted on the lowest priority class of service. At low bandwidth utilization, the high-fidelity high-bitrate packets will be used and during congestions, these packets would be dropped. This ensures achieving the best possible voice quality during varying network loads. Simulation results measuring the mean opinion score of voice quality show that superior voice quality can be achieved with this method over using standalone codecs while maintaining excellent quality of service for data traffic as well.

Effect of Codec Bit Rate and Packet Loss on Thai Speech Recognition over IP

Abstract: Nowadays, VoIP has become the core communications on the internet. One of the crucial applications on VoIP is the automated IVR system that interacts with the user automatically. Speech recognition plays an important role behind this kind of system. This paper studies the effect of codec bit rate and network packet loss on Thai speech recognition systems over an IP network. We encoded the speech samples of male, female and artificial voice with various bit rates of Speex codec. The speech sample was sent by RTP through the IP network with packet loss simulation. The speech quality was measured by PESQ and compared to word error rate of speech recognition. The results show that the codec bit rate and level of packet loss have a significant impact on the performance of speech recognition over IP.

RTP Payload Format for the Enhanced Variable Rate Narrowband-Wideband Codec (EVRC-NW)

Abstract: This document specifies Real-time Transport Protocol (RTP) payload formats to be used for the Enhanced Variable Rate Narrowband-Wideband Codec (EVRC-NW). Three media type registrations are included for EVRC- NW RTP payload formats. In addition, a file format is specified for transport of EVRC-NW speech data in storage mode applications such as email.

Quality Improvement of VoIP CODEC Using Tandem Technique

Abstract: Variety of compression algorithms of each VoIP CODEC makes the quality of the output files different. As voices may need to be encoded multiple times while being sent through multiple networks that use different CODEC (so called CODEC tandem), there is a possibility that the CODECs which provide low voice quality may be used together with others that provide much better voice quality. The main purposes of this research are to analyze and seek improvement in the quality of voices encoded using VoIP CODEC

The optimization of parameters configuration for AMR codec in mobile networks

Abstract: Although the pursuing of the data service diversity has become the mainstream, the speech quality is of vital importance for the network operator in the intensifying competition telecommunication market The Adaptive Multi-Rate (AMR) codec mode, which is one of the most significant techniques, is recommended by 3rd Generation Partnership Project (3GPP) for audio data compression Many studies relevant to AMR codec, including the parameter configuration, have been progressed However, most of them are studied by simulations In this paper, the study of the configuration of Threshold (THR), Hysteria (HYST) and Initial Codec Mode (ICM) for AMR-FR and AMR-HR are implemented in the commercial network Totally, 26 schemes are put forward and tested to obtain the optimization configuration, which will be used in the current network to improve the speech service quality

Adaptive multimedia codec parameter adaptation

Abstract: There is provided determining of multimedia codec parameters for a multimedia communications session. A channel quality indicator value representing current throughput characteristics of a communications channel used for the multimedia communications session is acquired. The channel quality indicator value is compared to a previous channel quality indicator value of the communications channel. It is determined, based on the comparing, whether to delay adaptation of values of a set of multimedia codec parameters to be used by a multimedia codec during the multimedia communications session or not, and which values of the set of multimedia codec parameters to adapt.