Showing papers on "Enhanced Variable Rate Codec published in 2005"

Method and system for terminal codec setup of multimedia ring back tone service

Abstract: Disclosed is a method for terminal codec setup for a multimedia Ring Back Tone (RBT) service, which allows RBT sound sources previously set in a sound source-providing server by a called subscriber to be reproduced to an originating terminal by means of a Home Location Register (HLR) and the sound source-providing server for storing the RBT sound sources when a calling subscriber telephones the called subscriber, the HLR storing profile information including whether the subscriber has joined the RBT service. The method includes the steps of : (a) receiving a first codec setup message including information (multimedia codec information) regarding a multimedia codec from the called subscriber, after transmitting an ISDN User Part (ISUP) call connection request message including the multimedia codec information to the called subscriber; (b) when the first codec setup message is received, transmitting a second codec setup message for requesting setup of the multimedia codec to a call-side Base Transceiver Station (BTS) , thereby controlling a call-side vocoder located in a call-side Base Station Controller (BSC) to set the multimedia codec; (c) when the first codec setup message is received, transmitting a third codec setup message for requesting setup of the multimedia codec to the originating terminal, thereby controlling the originating terminal to set the multimedia codec; and (d) receiving an RBT sound source selected using the multimedia codec information from the sound source- providing server and transmitting the RBT sound source to the originating terminal.

Interactive Codec Selection

Abstract: A codec selection process is independent of a codec source. A codec table may be used that includes indexed codec references related to codecs stored in a codec source. The codec table and the codec source may be modified without affecting a codec selection process. This feature of an exemplary implementation makes it fairly straightforward to add, change, or otherwise modify codecs stored in the codec source and referenced in the codec table without having to change the codec selection process.

Session data and setting method thereof in a sychronous wireless communication system

Abstract: In session data and a method for establishing a session in a synchronous wireless communication system in accordance with the present invention, by additionally defining a maximum bit rate parameter of the EVRC such that the maximum rate can be varied according to circumstances and conditions of a wireless network, the capacity of the wireless network is expanded and therefore more efficiently used. Services can be flexibly implemented according to the circumstances and conditions of the wireless network by using the maximum bit rate parameter for increasing or decreasing the bandwidth of the wireless network.

Transcoding method for switching between selectable mode voice encoder and an enhanced variable rate CODEC

Abstract: An apparatus comprising an encoder circuit and a transcoder circuit. The encoder circuit may be configured to generate a bitstream comprising a series of packets in response to a speech input signal. The transcoder circuit may be configured to generate an intermediate bitstream in response to the bitstream. The transcoder (a) implements (i) a first encoding type comprising a selectable mode voice (SMV) encoding or (ii) a second encoding type comprising an enhanced variable rate (EVR) encoding in response to a type of data in each of the packets of the bitstream and (b) the first or second encoding type is selected on a per packet basis.

Adaptive thresholds for AMR codec mode selection

Abstract: The speech codecs from the adaptive multi-rate (AMR) codec family enable provisioning of excellent speech quality, at the same time providing a way forward towards state-of-the-art, spectrally efficient, high capacity cellular networks. One straightforward way to characterize the benefit of AMR speech codecs is that the robustness to interference and noise in radio networks is increased and that this advantage over other, nonadaptive, speech codecs can be capitalized on in several different ways, e.g., by enhancing speech quality or improving system capacity. In this paper, improved mode adaptation, where codec mode switching thresholds are adaptive to radio conditions, is discussed. Example simulations show that an adaptive thresholds algorithm applied to GSM can significantly improve objective speech quality. Corresponding improvements were also found in informal listening tests.

Rate determination based on perceptual loudness

Abstract: We describe a perceptually-motivated rate determination algorithm (RDA) for variable bit rate speech coding. Unlike existing rate selection strategies that are based on a voice activity detector and energy thresholds, the proposed method employs a perceptual loudness (PL) measure. The TIA IS-127 enhanced variable rate codec (EVRC) has been chosen as the test-bed for evaluating the performance of the PL-based rate selection strategy relative to three existing methods. In particular, the comparative study includes the following rate determination algorithms: voice activity detection; speech frame energy-thresholding; phonetic segmentation. Experimental results demonstrate that the proposed PL-based RDA compares well against other rate selection techniques in terms of average bitrate and speech quality.

Method for codec mode adaptation of adaptive multi-rate codec regarding speech quality

Abstract: Method of codec mode adaptation of a speech codec in dependency of the prevailing channel condition for transmission of speech frames in a telecommunication system comprising the steps of determining a bit error rate (BER) from a estimated carrier to interferer ratio (C/I) per speech burst, generating a frame BER value of speech frame from a plurality of consecutive bursts and determining a critical BER level for a plurality of speech frames by a maximum operation of the frame BER values of the plurality of speech frames.

Codec switching device

Abstract: PROBLEM TO BE SOLVED: To make it impossible that a codec which has developed trouble or the like to become unable to transmit is judged as most suitable for transmission. SOLUTION: The codec switching device 10 includes a plurality of codecs a1 to an and a controller 13. The controller 13 searches its own codec information s1 containing a transmission band of the plurality of the codecs a1 to an and other codec information s2 containing a transmission band of a plurality of codecs b1 to bn of a transmission party as well as network information s3 containing a size of a free bandwidth of a network 30. Based on the searched results, a codec utilized at the time of transmission is selected from among the plurality of codecs a1 to an. On this occasion, a codec with the most large-sized transmission bandwidth is selected from among free bandwidths of the network 30. These operations are implemented also during transmission. COPYRIGHT: (C)2007,JPO&INPIT

Method for improving the quality of sound by using a dual codec in a subscriber-based ringback tone service, especially concerned with providing a specific ringback tone sound, registered by a called subscriber, to a calling subscriber with a more enhanced tone quality

Abstract: PURPOSE: A method for improving the quality of sound using a dual codec in a subscriber-based ringback tone service is provided to regenerate a ringback tone sound source at a better tone quality by releasing channel coding, such as an EVRC(Enhanced Variable Rate Codec) method, a QCELP(Qualcomm Code Excited Linear Prediction) method, etc., and executing sound source regeneration through a traffic channel suitably for the one-way transmission of a ringback tone sound source. CONSTITUTION: If a calling-side MSC(26) transmits a call connection request to a called-side MSC(22) as a calling mobile terminal(12) attempts call origination for a called mobile terminal(10), channel coding for a called-side BSC(18) is released. The called-side BSC(18) executes coding, suitable for the one-way transmission of ringback tone data, for a ringback tone sound source, registered by the subscriber of the called mobile terminal(10), and sends it to the calling mobile terminal(12). If the call connection between the two mobile terminals is made, the called-side BSC is transferred to a channel coding state so that a voice call can be initiated.

Apparatus and method for efficiently supporting voip in a wireless communication system

Abstract: The present invention concerns methods for efficiently supporting Voice over Internet Protocol (VoIP) on the Forward Packet Data Channel (F PDCH) in CDMA 2000 1xEV-DV systems. Active speech in VoIP is encoded using, for example enhanced variable rate codec (EVRC), which produces 171, 80 and 16 bits per 20 ms of speech for Rate 1, Rate %2 and Rate 1/8, respectively. However, about 60% of the time a user is inactive during a speech session, so an inordinate amount of system bandwidth is comprised of rate 1/8 VoIP packets. In one embodiment of the present invention the apparatus of the present invention identifies the Rate 1/8 voice frame packets and discards them. In another embodiment of the present invention, the apparatus of the present invention identifies the Rate 1/8 voice frame packets and selects some of them for further transmission. In both embodiments the efficiency of channel utilization is increased since the amount of channel band width used to communicate relatively little information, e.g., gaps of silence, is decreased.

Single-Microphone Noise Suppression for 3G Handsets Based on Weighted Noise Estimation

Abstract: A noise suppression algorithm with high speech quality based on weighted noise estimation is presented This algorithm continuously updates the estimated noise by weighted noisy speech in accordance with an estimated SNR With a better noise estimate, a more correct SNR is obtained, resulting in the enhanced speech with low distortion Subjective evaluation results show that five-grade mean opinion scores of this algorithm with a speech codec is improved by as much as 035, compared with either the MMSE-STSA or the EVRC noise suppression algorithm A noise suppressor based on a later version of this noise suppression algorithm satisfies all the 3GPP minimum performance requirements It is employed in the world’s first 3G handset equipped with a 3GPP-endorsed noise suppressor

Synchronizing a channel codec and vocoder of a mobile station

Abstract: In one embodiment, the present invention includes a method for maintaining a vocoder and channel codec in substantial synchronization. The method may include receiving a configuration message that includes rate information and an effective radio block identifier at a mobile station, coding a current radio block via a vocoder and channel codec, configuring an encoding portion of the vocoder and channel codec with the rate information after performing the coding, and then coding the effective radio block using the rate information. Other embodiments are described and claimed.

A fast encoding technique for vector quantization of LSF parameters

Abstract: A fast encoding technique is described for vector quantization (VQ) of line spectral frequency parameters. A reduction in VQ encoding complexity is achieved by using a preliminary test that reduces the necessary codebook search range. The test is performed based on two criteria. One criterion uses the distance between a specific single element of the input vector and the corresponding element of the codevectors in the codebook, The other criterion makes use of the ordering property of LSF parameters. The fast encoding technique is implemented in the enhanced variable rate codec (EVRC) encoding algorithm. Simulation results show that the average searching range of the codebook can be reduced by 44.50% for the EVRC without degradation of spectral distortion (SD).

A New Wideband Speech CODEC AMR-WB

Abstract: This paper presents a new speech CODEC which will be used in 3G mobile communication systems and packet networks. A bandwidth of 50 to 7000 Hz improves the intelligibility and naturalness of speech, so it is suitable for multimedia services of the third generation mobile communication systems, wideband telephony over packet networks, audio and video teleconferencing. After a brief discussion about the background of the CODEC, this paper gives a detailed description of its structure and characters. Because the AMR-WB is a new CODEC, and is passed in March 2001,the research about it is still in the initial phase and has not gotten into use. And the research about it has not been taken. So it is necessary to have a deep research in this area and it will be beneficial to the future study.

Magnitude-Sign split quantization for bandwidth scalable wideband speech codec

Abstract: New quantization method based on magnitude-sign split scheme for bandwidth scalable wideband speech codec is proposed. In the high-band codec, the signal is band-pass filtered and each band is transformed independently into DCT domain. The DCT coefficients are split into magnitude and sign, and each is quantized separately based on its unique characteristics. In addition, the quantized gain parameter in the low-band codec is utilized in the high-band codec for an enhanced performance. The 19.8kbps bandwidth scalable wideband codec consisting of G.729E for low-band and the proposed codec for high-band is developed, and it is confirmed that the proposed codec has better subjective performance than 24kbps G.722.1.

Software-Based video codec for mobile devices

Abstract: With the rapid development of wireless networks and consumer electronics, various mobile applications have emerged. However, due to some constraints such as weak computational power, limited memory and small display screen, traditional video coding applications can not work well on mobile devices. In this paper, we proposed a software-based video codec framework and its implementation which is suitable for real-time video coding applications on mobile devices. Some key optimizing techniques, such as fast predictive motion estimation (ME), zero-coefficients prejudgment and multiplierless integer discrete cosine transform (DCT), are used in our codec. Experimental results demonstrate the flexibility of our framework and the good speedup we achieved while video quality degradation is negligible. The codec is suitable for scenarios where low-complexity computing is required.

Fast Codec Design and Implementation using WHT and R-D Optimization in Wireless Handset

Abstract: An image compression codec based on WHT is proposed and im- plemented in wireless handset. Considering the low processing power of wireless handset, fast decoding codec is proposed and implemented. The proposed codec consists of RCT, WHT transform, quantization using R-D optimization and lossless coding. The test results for wireless handsets show that the proposed codec has a better performance than the IJG JPEG codec.

MPEG-2 real-time software CODEC for full-duplex transmission application over IP networks

Abstract: This paper proposes a real-time MPEG-2 software CODEC for full-duplex transmission applications, and evaluates its performance and usefulness. The CODEC consists of a high-speed encodersdecoder, an IP sendersreceiver, and an error recovery controller. Each encodersdecoder is accelerated and optimized by exploiting fast algorithms and instruction-level parallelism. The IP sendersreceiver combination achieves low delay owing to the direct translating of each elementary stream of video and audio into UDPsIP packets. The error recovery controller carries out simple but powerful error tolerance against packet loss over IP networks. This CODEC attains low delay of 99 ms (M = 1, N = 1) to 165 ms (M = 3, N = 3) including input, encoding, transmitting, decoding, and output delays, and maintains a normal frame rate of 30 fps (frames per second) and more than 20 fps even under a fairly heavy network load. It provides sufficiently good performance for use as a real-time full NTSC-size CODEC on a PC of at least 1.2-GHz CPU. © 2005 Wiley Periodicals, Inc. Syst Comp Jpn, 36(2): 33–41, 2005; Published online in Wiley InterScience (). DOI 10.1002sscj.20151

Software-based video codec for mobile devices

Abstract: With the rapid development of wireless networks and consumer electronics, various mobile applications have emerged. However, due to some constraints such as weak computational power, limited memory and small display screen, traditional video coding applications can not work well on mobile devices. In this paper, we proposed a software-based video codec framework and its implementation which is suitable for real-time video coding applications on mobile devices. Some key optimizing techniques, such as fast predictive motion estimation (ME), zero-coefficients prejudgment and multiplierless integer discrete cosine transform (DCT), are used in our codec. Experimental results demonstrate the flexibility of our framework and the good speedup we achieved while video quality degradation is negligible. The codec is suitable for scenarios where low-complexity computing is required.

Embedding Perceptual Metrics in Rate Control Algorithms

Abstract: In this paper, we describe a perceptually-motivated rate control algorithm for variable bit rate (VBR) speech coding. Unlike the existing rate selection strategies that are based on statistical metrics, the proposed method employs a perceptual loudness (PL) measure. The VBR speech standard TIA IS-127 enhanced variable rate codec (EVRC) has been chosen as the test-bed for evaluating the performance of the PL-based rate selection strategy relative to three other existing methods. In particular, the comparative study includes the following rate determination algorithms: voice activity detection, speech frame energy-thresholding, and phonetic segmentation. Experimental results demonstrate that the proposed PL-based RDA compares well against other rate selection techniques in terms of average bitrate and speech quality

High-Band Codec for Bandwidth Scalable Wideband Speech Codec

Abstract: In this paper, the high-band codec for bandwidth scalable wideband speech codec is proposed. The wideband input speech signal is separated into low-band signal and high-band signal, and the low-band signal is encoded by the standard narrow-band speech codec and the high-band signal is encoded by the proposed codec. In the high-band codec. the signal is transformed into frequency domain by MLT on a subframe basis, and MLT coefficients are splitted into magnitude and sign for quantization. The magnitudes of MLT coefficients are arranged into several time-frequency bands and each band is quantized in 2D-DCT domain, where the low-band information is utilized for better performance. The sign of MLT coefficient is quantized based on a priority selection process with the weighting measurement. The objective and subjective performance of wideband speech codec including the proposed high-band codec is measured, and it is confirmed that the proposed codec has better performance than 32kbps G.722.1.