Showing papers on "Enhanced Variable Rate Codec published in 2012"

Non intrusive codec identification algorithm

Abstract: We present a non-intrusive data driven method for codec detection and identification in the presence of background noise. The method uses a number of speech features which are then used to train a CART classifier. We demonstrate the performance of the method using several different noise types over a wide range of SNRs. Our results show that we can identify a codec and its bit rate to an accuracy of 92% and we are able to detect the presence of a codec with an accuracy of 97% at −5 dB SNR.

Dynamic video switching

Abstract: In an example, a dynamic codec allocation method is provided. The method includes receiving a plurality of datastreams and determining a respective codec loading factor for each of the datastreams. The datastreams are assigned to codecs, in order by respective codec loading factor, starting with the highest respective codec loading factor. Initially, the datastreams are assigned to a hardware codec, until the hardware codec is loaded to substantially maximum capacity. If the hardware codec is loaded to substantially maximum capacity, the remaining datastreams are assigned to a software codec. As new datastreams are received, the method repeats, and previously-assigned datastreams can be reassigned from a hardware codec to a software codec, and vice versa, based on their relative codec loading factors.

An Adaptive Codec Switching Scheme for SIP-Based VoIP

Abstract: Contemporary Voice-Over-IP (VoIP) systems typically negotiate only one codec for the entire VoIP session life time. However, as different codecs perform differently well under certain network conditions like delay, jitter or packet loss, this can lead to a reduction of quality if those conditions change during the call. This paper makes two core contributions: First, we compare the speech quality of a set of standard VoIP codecs given different network conditions. Second, we propose an adaptive end-to-end based codec switching scheme that fully conforms to the SIP standard. Our evaluation with a real-world prototype based on Linphone shows that our codec switching scheme adapts well to changing network conditions, improving overall speech quality.

Very low-delay H.264 codec for consumer applications

Abstract: We have newly developed a very low-delay full HD codec with relatively lower bitrate focusing especially on consumer or small business applications. We have analyzed several important factors for achieving low-delay, and developed a new codec pipeline control scheme for our versatile H.264 codec platform already developed. We implemented the algorithm on the codec platform, and obtained a result of 10 ms minimum delay with several test sequences at bitrate of 8 to 10 Mbps.

On the performance of IP and mobile based Automatic Speaker Verification

Abstract: Automatic Speaker Verification (ASV) is a challenging task over the mobile/IP based system as the coding introduces some loss in system performance This paper reports on the work in progress to examine the impact of GSM-AMR codec used in mobile at its various bit rates and G729 codec for VoIP, along with different kind of noise and packet loss scenario for the speech signal PURE YOHO database has been used for the evaluation of this task Respective encoder and decoders are used back to back on wideband clean microphone speech to simulate the real-life situation Evaluation of performance is done through the measurement of Equal Error Rate (EER) It has been shown that the loss due to GSM-AMR codec is very significant for speaker verification compared to undecoded speech Though the packet loss and bit rate may degrade the quality of speech but it is not significant to detection of speaker's identity

Embedded bandwidth scalable wideband codec using hybrid matching pursuit harmonic/CELP scheme

Abstract: Recent advances in speech coding have made wideband coding feasible at the bit-rates sufficient for mobile communication. Here we propose a novel hybrid harmocic Code Excited Linear Prediction (CELP) scheme for highband coding of band-split scalable wideband codec, where the low-band (0---4 kHz) is critically subsampled and coded selectively using existing narrowband codecs such as 5.4 kbps and 6.3 kbps G.723.1, 8 kbps G.729, and 11.8 kbps G.729E. The high-band signal is divided into stationary mode (SM) and non-stationary mode (NSM) components based on its unique characteristics. In the SM portion, the high-band signal is compressed using a multi-stage coding that combines the sinusoidal model and CELP. The first stage coding applies the damping factor matching pursuit (MP) algorithm without either the Over-Lap-Add (OLA) or smoothly interpolative synthesis schemes and the second stage utilizes CELP with the circular codebook. In the NSM portion, the high-band signals are coded by CELP with both pulse and circular codebooks by applying the complexity-reduced algorithm. To ensure scalability in highband coding, two enhancement layers are used to increase the number of pulses and control the quantizing sinusoidal parameter numbers. This paper describes the new algorithm and discuses novel techniques for efficient bandwidth wideband speech coding and subjective quality performance. For efficient bit allocation and enhanced performance, the pitch of the high-band codec is estimated using the quantized pitch parameter in low-band codec. An informal listening test, rated the subjective speech quality as comparable to that obtainable with G.722.2 as the fullband wideband codec and G.722.2 as the highband codec, the recent standardized band-split wideband codec.

An Investigation into the Performance Enhancement and Salient Features of AMR-WB Speech Codec Used in GSM Communication

Abstract: The 3rd Generation Partnership Project (3GPP) along with European Telecommunication Standard Institute (ETSI) recommended new speech coding scheme, called Adaptive Multi Rate (AMR) in 2000 and 2001. The proposed scheme has two variants: AMR-Narrowband (AMR-NB) and AMR-WB (AMR- Wideband). The speech quality of AMR-WB in GSM systems is remarkably high compared with that of AMR-NB speech codec because it encompasses wider bandwidth of the speech signal, 50-7000 Hz, instead of 200-3400 Hz in AMR-NB. But there are always possibilities for improvement. This paper investigates the salient features of AMR-WM codec and various possibilities for further enhancement in its speech quality. Two algorithms are recommended for incorporation into the standard AMR-WB to achieve this goal.

Carving deleted voice data in mobile

Abstract: People leave voicemails or record phone conversations in their daily cell phone use. Sometimes important voice data is deleted by the user accidently, or purposely to cover up criminal activity. In these cases, deleted voice data must be able to be recovered for forensics, since the voice data can be used as evidence in a criminal case. Because cell phones store data that is easily fragmented in flash memory, voice data recovery is very difficult. However, if there are identifiable patterns for the deleted voice data, we can recover a significant amount of it by researching images of it. There are several types of voice data, such as QCP, AMR, MP4, etc.. This study researches the data recovery solutions for EVRC codec and AMR codec in QCP file, Qualcumm's voice data format in cell phone.