Effects of PRMA on objective speech quality

When transmitting 32 kbit/s adaptive differential pulse code modulation (ADPCM) speech using Reed-Solomon error correction coding and 16 level quadrature amplitude modulation (16-QAM), a 20 slot packet reservation multiple access (PRMA) assisted cordless telecommunications (CT) scheme supported 36-38 speech users with negligible objective and subjective speech degradation. The average number of users per slot was nearly doubled due to deploying PRMA and toll quality speech was transmitted in a user bandwidth approximately 11.6 kHz. For a channel signal-to-noise ratio (SNR) in excess of 25 dB, a Rayleigh fading channel and mobile speeds above 2 mph the speech segmental SNR degradation was less than 0.3 dB.<>     

An intelligent multimode voice communications system for indoorcommunications

A novel high-quality, low-complexity dual-rate 4.7 and 6.5 kbits/s algebraic code excited linear predictive codec is proposed for adaptive multi-mode speech communicators, which can drop their source rate and speech quality under network control in order to invoke a more error resilient modem amongst less favorable channel conditions. Source-matched binary Bose-Chaudhuri-Hocquenghem (BCH) codecs combined with unequal protection diversity- and pilot-assisted 16and 64-level quadrature amplitude modulation (16-QAM, 64-QAM) are employed in order to accommodate both the 4.7 and the 6.5 kbits/s coded speech bits at a signaling rate of 3.1 kBd. Assuming an excess bandwidth of 100%, in a bandwidth of 200 kHz 32 time slots can be created, which allows us to support in excess of 50 users, when employing packet reservation multiple access (PRMA). Good communications quality speech is delivered in an equivalent bandwidth of 4 kHz, if the channel signal-to-noise ratio (SNR) and signal-to-interference ratio (SIR) of the benign indoors cordless channel are in excess of about 15 and 25 dB for the lower and higher speech quality 16-QAM and 64-QAM systems, respectively, and the PRMA time-slots are sufficiently uninterfered due to using time-slot classification algorithms and due to the attenuation of partitioning walls and ceilings     

A low-rate multi-level voice/video transceiver for personal communications

A personal communication system (PCS) transceiver is proposed and investigated. A 4.8 kbit/s transformed binary pulse excited (TBPE) linear predictive speech codec, embedded source sensitivity-matched binary Bose-Chaudhuri-Hocquenghem (BCH) block error correction codecs, non-coherent differentially coded 16-level quadrature amplitude modulation (16-QAM) modem and packet reservation multiple access (PRMA) are deployed. The 2.15 kBd transceiver requires a signal-to-noise ratio (SNR) and signal-to-interference ratio (SIR) in excess of about 24 dB over Rayleigh-fading channels in order to support 10–11 nearly un-impaired voice conversations within a bandwidth of 30 kHz. Additionally, by reserving two PRMA time slots for video telephony, an 8.52 kbps videophone user can also be supported.The constructive criticism of the anonymous Reviewers is gratefully acknowledged.     

A subband coding, BCH coding, and 16-QAM system for mobile radiospeech communications

A combined subband speech coding (SBC), Bose-Chaudhuri-Hocquenghem (BCH) error-correction coding, and 16-level quadrature amplitude modulation (16-QAM) scheme with switched diversity and speech postenhancement is proposed. The system's performance is dramatically improved by deploying some degree of fade tracking capability over fading channels. Further quality enhancement accrues by using appropriate mapping between the SBC speech codec and the Gray coded QAM words. Various BCH codes are utilized to adequately match the error-correcting power to the perceptual importance of the SBC bits. One of the proposed systems operates at 7 kBd and yields good communications-quality speech for channel signal-to-noise ratios (SNRs) in excess of 20 dB and encounters a maximum overall system delay of 55.125 ms. A more complex arrangement uses second-order switched diversity to reduce the channel SNR required to around 16 dB and the transmission rate to 5 kBd when the vehicular speed is 30 mph while the system delay is unchanged at 55.125 ms     

ADPCM with a multiquantizer for speech coding

A speech coding algorithm with low complexity and a short processing delay is introduced. The proposed algorithm is ADPCM (adaptive digital pulse code modulation) with a multiquantizer (ADPCM-MQ). The input signal is processed in parallel by multiple ADPCM coders with different characteristics. Then the optimum ADPCM coder with minimum error power is dynamically selected for each frame. A 16-kb/s codec based on this algorithm has been implemented using two general-purpose digital signal processors (MB8764) with 8.3 ms of total processing delay. A segmental SNR of 19-21 dB was achieved at 16 kb/s; with postfiltering the segmental SNR was increased to 23-25 dB. Combined with the time domain compression scheme, the algorithm can be easily applied to 8-kb/s coding. It is also extensible to variable-rate coding     

Application of NB/WB AMR speech codecs in the 30-kHz TDMA system

A new system enhancement method is proposed for the EIA/TIA-136 system offering both channel operational range extension and improved performance within the current operational range. The existing time-division multiple-access (TDMA) (136) speech codec, the IS-641 enhanced full rate vocoder, operates at a fixed bit rate and does not allow the reallocation of bits to channel error protection as channel conditions degrade. The research presented here investigates the application of the narrow-band adaptive multirate (NB-AMR) speech codec and the wide-band AMR (WB-AMR) codec, both originally designed for the 200 kHz GSM channel, in the TDMA (TIA/EIA-136) 30-kHz system. In particular, we investigate adaptively allocating bits between NB/WB speech coding and error control coding within the limited channel bandwidth. Four modes out of 17 have been carefully chosen for the new TDMA/AMR system. Switching between codec rates as channel conditions change produces range extension below a C/I of 15 dB while also improving performance in the existing operational range above 15 dB. We keep the time slot formats unchanged so that our method is completely compatible with existing 136 systems.     

A packet-based CAPDM speech coder for PCN applications

In this paper, we present a median-rate speech coder, the controlled adaptive prediction delta modulation coder (CAPDM), which operates at 16 kb/s with good speech quality and low algorithm complexity. The coder is dedicated to personal communication network (PCN) applications and transmits speech samples on the basis of packets. It combines the features of a one-step looking forward decision, syllabic companding, instantaneous companding, and adaptive prediction. In addition to the use of a short-term prediction filter, CAPDM also exploits the pitch property to predict speech waveform explicitly. With the aid of a pitch prediction filter, the performance of a CAPDM codec improves about 3 dB in segmental signal-to-noise ratio (SEGSNR). The average SEGSNR of CAPDM.FF is about 21 dB, which is 7 dB over traditional CVSD at 16 kb/s. We also utilize an adaptive postfilter (APF) to enhance the perceptual quality of the decoded speech. The mean opinion score (MOS) listening test of CAPDM.FF with APF shows that its average score achieves 4.19, which is as good as G.728 16-kb/s LD-CELP and is comparable with CCITT G.721 32-kb/s ADPCM. The complexity of CAPDM.FF is evaluated to be 8 MIPS, which is much lower than that of LD-CELP and could be further reduced by adopting a smaller correlation window for pitch detection. To solve the problem of packet loss, we developed a packet-based waveform substitution method by reinitializing the codec parameters at the beginning of each packet. The simulation results show that CAPDM.FF could tolerate 5% of packet loss and still keep an SEGSNR at 10 dB and an MOS at about 3.0     

Shared time division duplexing: an approach to low-delayhigh-quality wireless digital speech communications

Various strategies to provide low-delay high-quality digital speech communications in a high-capacity wireless network are examined. Various multiple access schemes based on time-division and packet reservation are compared in terms of their statistical multiplexing capabilities, sensitivity to speech packet dropping, delay, robustness to lossy packet environments, and overhead efficiency. In particular, a low-delay multiple access scheme, called shared time-division duplexing (STDD) is proposed. This scheme allows both the uplink and downlink traffic to share a common channel, thereby achieving high statistical multiplexing gain even with a low population of simultaneous conversations. The authors also propose a choice of low delay, high quality speech coding and digital modulation systems based on adaptive DPCM, with QDPSK or pseudo-analog transmission (skewed DPSK), for use in conjunction with the STDD multiple access protocol. The choice of the alternative systems depends on required end-to-end delay, recovered speech quality and bandwidth efficiency. Typically, with a total capacity of 1 MBaud, 2 ms frame and 8 kBaud speech coding rate, low delay STDD is able to support 48 pairs of users compared to 38, 35, and 16 for TDMA with speech activity detection, basic TDMA and PRMA respectively. This corresponds to respective gains of 26%, 37% and 200%     

Packet reservation multiple access for wireless multimedia communications

Packet reservation multiple access (PRMA) is portrayed as a multimedia packet multiplexer conveying speech, data and video signals, which ensures that the slot occupancy of conventional time division multiple access (TDMA) links is approximately doubled. In addition to 20 speech users the 20 slot 720 kbit/s scheme presented supported 20 data users and up to seven video users, while maintaining a slot occupancy in excess of 80%.<>     

OFDM-based turbo-coded hierarchical and non-hierarchicalterrestrial mobile digital video broadcasting

The feasibility of terrestrial digital video broadcast (DVB) to mobile receivers is studied and turbo coded performance enhancements are proposed. Initially, the MPEG-2 codec is subjected to a rigorous bit error sensitivity investigation, in order to assist in designing various error protection schemes for wireless DVB transmission. The turbo codec is shown to provide signal-to-noise ratio (SNR) performance advantages in excess of 5-6 dB over conventional convolutional coding both in terms of bit error rate and video quality. Our experiments suggested that-despite our expectations-multi-class data partitioning did not result in error resilience improvements, since a high proportion of relatively sensitive video bits had to be relegated to the lower integrity subchannel, when invoking a powerful low-rate channel codec in the high-integrity protection class. Nonetheless, DVB transmission to mobile receivers is feasible, when using turbo-coded OFDM transceivers at realistic power-budget requirements under the investigated highly dispersive fading channel conditions. It is interesting to note furthermore that the 5-6 dB SNR improvement due to turbo coding allows us to invoke for example the double-throughput 16-level quadrature amplitude modulation (16-QAM) mode instead of the standard convolutional-coded 4-QAM mode. This facilitates doubling the bit rate and hence improving the video quality     

Quality of service and bandwidth efficiency of cellular mobile radio with variable bit-rate speech transmission

The performance of an adjustable source/channel codec in a cellular mobile-radio environment is investigated. The speech transmission rate and the amount of forward error correction change in response to changing channel conditions. The channel rate is constant at 32 kb/s, and when the channel is good all of these bits are used for speech transmission. In intermediate and poor channels the speech rate is 24 or 16 kb/s, and the remaining channel symbols are used for forward error correction. Relative to conventional transmission this approach offers an improved grade of service. For example, the outage rate (the proportion of "poor or worse" communications) goes from nine percent with fixed-rate to three percent with variable-rate transmission. Alternatively, this improved grade of service can be exchanged for higher bandwidth efficiency. The fixed-rate system (with nine percent outage) has 23 users per cell. With 52 users per cell the outage of the variable-rate system is only six percent.     

Dual-mode vector-quantized low-rate cordless videophone systems forindoors and outdoors applications

Dual-mode reconfigurable wireless videophone transceivers are proposed for noise-, rather than interference-limited indoors and outdoors applications and their video quality, bit rate, robustness, and complexity issues are analyzed. A suite of fixed, but arbitrarily programmable low-rate, perceptually weighted vector quantized (VQ) codecs with and without run-length compression (RLC) are contrived for quarter common intermediate format (QCIF) videophone sequences. The 11.36-kb/s Codec 1 is Bose-Chaudhuri-Hochquenghem (BCH) (127,71,9) coded to a rate of 20.32 kb/s and this arrangement is comparatively studied along with the 8-kb/s Codec 2 and BCH (127,50,13) scheme, which has the same 20.32-kb/s overall rate. The source-sensitivity matched Systems 1-6 characterized in a table were contrived to comparatively study the range of system design options. For example, using Codec 1 in System 1 and coherent pilot symbol assisted 16-level quadrature amplitude modulation (16-PSAQAM), an overall signaling rate of 9 kBd was yielded, if the noise-limited channel had a signal-to-noise ratio (SNR) in excess of about 22 dB in the vicinity of the basestation or in indoors scenarios. In contrast, over lower quality outdoors channels near the fringes of the cell, the more robust 4-QAM mode of operation had to be invoked, which required twice as many time slots to accommodate the resulting 18-kBd stream and hence, reduced the total number of users supported. The robustness of Systems 2-4, and 6 was increased using automatic repeat requests (ARQ), again, inevitably reducing the number of users supported, which was between 6 and 16. In a bandwidth of 200 kHz, similarly to the Pan-European GSM mobile radio system's speech channel, using Systems 1, 3, 4, or 5, for example, 16 and eight videophone users can be supported in the 16- and 4-QAM modes, respectively, while in dual-mode cells the number of users is between eight and 16. The basic system characteristics are highlighted     

话音活动检测的模型及其在移动通信中的应用

本文给出了基于ＡＤＰＣＭ编码的话音活动检测器及其模型，并应用该模型分析了分组预约多址协议的性能。     

A Bandwidth Reservation Multiple Access Protocol for Wireless ATM Local Networks

A bandwidth reservation multiple access scheme(BRMA) is proposed to resolve contention and assignbandwidth among multiple users trying to gain access toa common channel such as in mobile users contending for resources in an ATM-based cellular networkor a wireless local area network (LAN) with shortpropagation delays. The protocol is best suited tosupport variable-bit-rate (VBR) traffic that exhibits high temporal fluctuations. Each mobile user isconnected end-to-end to another user over virtualchannels via the base station that is connected to thewired ATM B-ISDN network. The channel capacity is modeled as a time frame with a fixed duration.Each frame starts with minislots, to resolve contentionand reserve bandwidth, followed by data-transmissionslots. Every contending user places a request for data slots in one of the minislots. If therequest is granted by the base station through adownlink broadcast channel, the user then startstransmission in the assigned slot(s). The number ofassigned slots varies according to the required qualityof service (QoS), such as delay and packet lossprobability. A speech activity detector is utilized inorder to indicate the talkspurts to avoid wastingbandwidth. Due to its asynchronous nature, BRMA is ratherinsensitive to the burstiness of the traffic. Since theassignment of the minislots is deterministic, therequest channels are contention-free and the data channels are collision-free. Hence, in spite ofthe overhead (minislots) in each frame, BRMA provideshigher throughput than Packet Reservation MultipleAccess (PRMA) for the same QoS, especially for high-speed systems. A better delay performance is alsoachieved for data traffic compared to Slotted Alohareservation-type protocol PRMA. In addition, BRMAperforms better in terms of bandwidth efficiency thanthe conventional TDMA or the Dynamic TDMA, wherespeech activity detectors are very difficult toimplement.     

APC-AB codec modules operating at 16 and 8 kbit/s

High-quality speech codec modules operating at 16 and 8 kb/s have been developed using an adaptive predictive coding with adaptive bit allocation (APC-AB) scheme. An optimized APC-AB algorithm is studied that reduces processing complexity while maintaining speech quality. The coding algorithm is implemented in two digital signal processors (DSPs). The DSP chips, a framing LSI circuit, a PCM codec, and some peripheral ICs are integrated in each of two compact packages, i.e. codec modules, operating at 16 or 8 kb/s. The codec module size is as small as 80 mm×50 mm×12 mm, and its typical power consumption is 500 mW using 2-μm CMOS LSI technology. At 16 kb/s this APC-AB codec achieves high speech quality, close to that of a 7-bit μ-law PCM. The codec modules are expected to be used for various applications such as customer premises multiplexers for digital leased lines, digital mobile radio, and stored-and-forward-message systems (voice-mail systems)     

A monolithic ΣΔ A/D and D/A converter with filter forbroad-band speech coding

The authors present a 5-V-only 14-b, 16 ksamples/s linear codec suitable as the audio part of a CCITT G722 codec. The device uses second-order sigma-delta modulation for both analog/digital (A/D) and digital/analog (D/A) conversion at 2.048 Msamples/s. A time-continuous modulator with integrated antialias filtering is used at the A/D side, obviating the need for an external antialiasing filter. The digital filters for decimation and interpolation are implemented with both a custom digital signal processor (DSP) and specialized hardware. The device was realized with 74000 transistors on a 31-mm² die in a 3-μm SACMOS technology. A dynamic range of more than 80 dB and a passband ripple of 0.3 dB were attained with A/D and D/A paths in cascade     

Pseudo-analog speech transmission in mobile radio communicationsystems

A low-complexity pseudo-analog speech transmission scheme is proposed for portable communications. It uses a speech coder based on adaptive differential pulse code modulation (ADPCM) in combination with a multilevel digital modulation technique such as M-ary DPSK or M-ary FSK and features low quantization noise, bandwidth efficiency, and robustness to transmission errors. A nonsymmetric M -ary DPSK scheme called skewed M-ary DPSK is proposed to enhance the noisy channel performance. Comparison to conventional analog FM and a digital speech transmission scheme using adaptive predictive coding and forward error correction (FEC) based on convolutional coding shows that the pseudo-analog system has the best objective signal-to-noise ratio performance under most channel conditions. Informal subjective evaluations rate the digital system superior to the pseudo-analog scheme for bad channels and conversely for good channels. It is concluded that the pseudo-analog system can be designed with low delay and high speech quality for good channels with high spectral efficiency     

Performance simulation of NC-PRMA with signatures in microcellular communication systems

Packet reservation multiple access (PRMA) improves capacity in microcellular systems compared with time division multiple access (TDMA) or frequency division multiple access (FDMA). In PRMA, when a mobile terminal has information packets to transmit, it contends with other terminals for access to a common radio channel. Therefore the main performance degradation is due to the collision of terminals simultaneously transmitting packets. In this paper we propose a non-collision PRMA (NC-PRMA) protocol with signatures to achieve a better performance than PRMA does. Two classes of duplexing schemes, frequency division duplexing (FDD) and shared time division duplexing (STDD), are explored and two speech activity models, slow and fast, in both FDD and STDD schemes are studied. From the results of a computer simulation it is observed that, with the constraint of a packet-dropping rate no greater than 0·01, NC-PRMA can support 38 (43) and 45 (49) users respectively under the FDD and STDD schemes if the slow (fast) speech activity detector is adopted. © 1998 John Wiley & Sons, Ltd.     

The Residual Encoder--An Improved ADPCM System for Speech Digitization

An improved system for speech digitization using adaptive differential pulse-code modulation (ADPCM) is described. The system uses an adaptive predictor, an adaptive quantizer, and a variable length source coding scheme to achieve a 4-5 dB increase in signal-to-noise ratio over previous ADPCM. The increase can be used to improve speech quality at moderate data rates on the order of 16 kbits/s or to retain the same quality and reduce the data rate to 9.6 kbits/s. The latter alternative permits the use of narrow-band channels. The implementation complexity is on the same order as other ADPCM systems.     

VoIP Scheduling Algorithm for AMR Speech Codec in IEEE 802.16e/m System

This letter proposes an efficient uplink scheduling algorithm for voice over Internet protocol (VoIP) services with adaptive multi-rate (AMR) speech codec in IEEE 802.16e/m systems. The proposed scheduling algorithm adopts the random access scheme during silent-period to reduce the waste of uplink bandwidth considering the characteristics of AMR speech codec. The numerical results show that the proposed algorithm can increase the maximum supportable number of voice users by 26% compared to the conventional extended real-time polling service (ertPS).