Packets of real-time information are sent with a source rate greater than zero kilobits per second, and a time or path or combined time/path diversity rate initially being zero kilobits per second. This results in a quality of service QoS, optionally measured at the sender or the receiver. When the QoS is on an unacceptable side of a threshold of acceptability, the sender sends diversity packets at an increased rate. Increasing the diversity rate while either reducing or maintaining the overall transmission rate is new. CELP-based multiple-description data partitioning sends the base or important information plus a subset of fixed excitation in one packet and sends the base or important information plus the complementary subset of fixed excitation in another packet. Reconstruction produces acceptable quality when only one of the two packets is received and better quality when both packets are received. Reconstruction provides for single and multiple lost packets.

Systems, processes and integrated circuits for rate and/or diversity adaptation for packet communications

In this paper, a variational mode decomposition (VMD) based method has been proposed for the instantaneous detection of voiced/non-voiced (V/NV) regions in the speech signals. In the proposed method, the VMD is applied in iterative way with specific input parameters. Firstly, the VMD decomposes the speech signal into two components, then, the VMD is applied successively on one of these two components based on suitably defined convergence criteria. It has been shown that the VMD applied in iterative way behaves as a low-pass filter and after convergence it provides separation of the fundamental frequency (F0) component from the speech signal. The envelope of the F0 component of the speech signal has been obtained using an analytical model based on single degree of freedom (SDOF). Automatic threshold has been computed from the obtained envelope in order to detect the V/NV regions in speech signals. The proposed method has been studied on speech signals and the corresponding electroglottograph (EGG) signals from the CMU-Arctic database in different noise conditions obtained from the NOISEX-92 database. Experimental results at various signal to noise ratios (SNRs) are included in order to show the effectiveness of the proposed method compared to the other existing methods for V/NV detection in speech signals.

Instantaneous voiced/non-voiced detection in speech signals based on variational mode decomposition

A method of reducing channel state information fed back in an adaptive orthogonal frequency division multiple access system when transmitting and receiving data packets between a base station and a mobile station using an orthogonal frequency division multiplexing method comprising adaptive modulation and coding. The method includes: transmitting a data packet comprising information as to whether adaptive modulation and coding is on or off from the base station to the mobile station via a downlink; and tracking a modulation and coding scheme level only when it is determined that adaptive modulation and coding is performed based on the information as to whether adaptive modulation and coding is on or off and feeding back the channel state information obtained from the tracking result from the mobile station to the base station via an uplink.

Method of reducing feedback channel state information within adaptive OFDMA system and adaptive OFDMA system using the same

In this chapter, we present an overview of the most widely used algorithms, standards, and applications of wideband and narrowband speech coding. Algorithms for speech coding are classified into four broad headings: (1) waveform coding techniques (including PCM, companded PCM, and DPCM), which are typically used for landline telephony, internet telephony, and secure military communications; (2) subband coding, including perceptually transparent multi-rate and embedded coding which is mainly used for internet and digital audio applications; (3) linear predictive analysis by synthesis coding (LPC-AS) algorithms, including multipulse LPC, CELP, SELP, VSELP, and low-delay CELP, which are typically used for digital cellular and telephony; and (4) LPC vocoders, including advanced vocoder algorithms (e.g., MELP, MBE, and PWI) are used for applications such as secure telephony and satellite telephony. Applications in areas such as voiceover IP (VoIP) and digital cellular are emerging and require a speech coder to gracefully adapt to rapidly changing channel conditions—a need that is met by embedded and multirate speech coders associated with joint source-channel coding algorithms. Measures of speech coder perceptual quality include subjective measures of intelligibility (DRT and DALT) and naturalness (MOS and DAM), as well as objective measures such as segmental SNR, Bark spectral distortion, PSQM, and PESQ. Speech coding standards are set by organizations including the ITU (for landline telephony), MPEG (for multimedia applications), ETSI (for European digital cellular), TIA (for U.S. digital cellular), and DDVPC (for United States military applications).


Keywords:

speech coding;
PCM;
subband coding;
CELP;
LPC;
digital cellular;
multimedia;
voiceover IP;
mean opinion score (MOS)

/pdf/speech-coding-fundamentals-and-applications-hutobo87n0.pdf

Speech Coding: Fundamentals and Applications

T A method for communication includes receiving first and second data frames (90, 94) over first and second communication links, respectively, the first and second data frames containing respective first and second replicas of data, which has been encoded with a Forward Error Correction (FEC) code. The FEC code in the received first and second data frames is decoded, and respective first and second soft quality ranks of the first and second data frames are computed based on the decoded FEC code. One of the first and second replicas of the data are selected based on the first and second soft quality ranks. The selected one of the first and second replicas of the data is provided as output (98).

Protected communication link with improved protection indication

We have developed an adaptive multi-rate (AMR) speech coder designed to operate under the GSM digital cellular full rate (22.8 kb/s) and half rate (11.4 kb/s) channels and to maintain high quality in the presence of highly varying background noise and channel conditions. Within each total rate, several codec modes with different source/channel bit rate allocations are used. The speech coders in each codec mode are based on the CELP algorithm operating at rates ranging from 11.85 kb/s down to 5.15 kb/s, where the lowest rate coder is a source controlled multi-modal speech coder. The decoders monitor the channel quality at both ends of the wireless link using the soft values for the received bits and assist the base station in selecting the codec mode that is appropriate for a given channel condition. The coder was submitted to the GSM AMR standardization competition and met the qualification requirements in an independent formal MOS test.

/pdf/an-adaptive-multi-rate-speech-coder-for-digital-cellular-1zahdo4l8k.pdf

An adaptive multi-rate speech coder for digital cellular telephony

Diversity schemes include information about packet n in future packets or send information about packet n via separate paths. If packet n is lost, it is reconstructed from information included in future packets or information received via separate paths. This paper presents CELP-based diversity schemes for voice over packet applications. The diversity schemes reduce the impact of packet losses while being efficient in terms of both bandwidth requirement and computational complexity. With our diversity schemes, transmission schemes that allocate bandwidth resources among diversity stages during congestion give significantly better performance than schemes that use no diversity during congestion, for the same bandwidth usage.

A.K. Anandakumar

Papers

An adaptive multi-rate speech coder for digital cellular telephony

Efficient CELP-based diversity schemes for VoIP