Silence insertion descriptor (sid) frame detection with human auditory perception compensation
    21.
    发明公开
    Silence insertion descriptor (sid) frame detection with human auditory perception compensation 审中-公开
    打破信号帧的测定人的听觉补偿描述(SID)

    公开(公告)号:EP1229520A3

    公开(公告)日:2004-01-21

    申请号:EP01000577.5

    申请日:2001-10-29

    IPC分类号: G10L19/00

    CPC分类号: G10L19/012

    摘要: A method to reduce the amount of bandwidth used in the transmission of digitized voice packets is described. The method is used to reduce the number of transmitted packets by suspending transmission during periods of silence or when only noise is present. The system determines if a background noise update is warranted based on human auditory perception factors instead of an artificial limiter on excessive silence insertion descriptor packets. The system searches for characteristics in the perceptual changes of background noise instead of analyzing speech for improved audio compression. The invention weighs factors affecting the perception of sound including frequency masking, temporal masking, loudness perception based on tone, and auditory perception differential based on tone.

    Constellation design for PCM upstream modulation
    22.
    发明公开
    Constellation design for PCM upstream modulation 审中-公开
    KonstellationsentwurffürPCM-Aufwärtsmodulation

    公开(公告)号:EP1363431A2

    公开(公告)日:2003-11-19

    申请号:EP03100951.7

    申请日:2003-04-08

    IPC分类号: H04L25/49 H04L1/00

    CPC分类号: H04L25/4927 H04L1/0003

    摘要: A method for constellation design in a telecommunications network (10) using pulse code modulation to transmit data signals upstream (30) between client and server voice-band modems (12,26). The invention selects a constellation for transmission over an analog channel of an equivalence class of data points using pulse code modulation based on the presence or absence of robbed bit signaling and interference from echo levels. The constellation is designed by determining noise in a PCM channel using a cumulative distribution function for echo (90), determining the extent of said noise for an array of possible constellation points (92), and selecting constellation points such that the largest negative noise of a first point remains above the largest positive noise level of a neighboring second point (94). Constellations are created off-line and stored (96) for retrieval during modulation depending on the level of the echo.

    摘要翻译: 一种用于在电信网络(10)中使用脉码调制以在客户端和服务器语音频带调制解调器(12,26)之间上行传输数据信号(30)的星座设计的方法。 本发明通过使用脉冲编码调制基于有无位带信令的存在或不存在以及来自回声电平的干扰来选择用于在等效数据类的模拟信道上进行传输的星座。 通过使用回波(90)的累积分布函数确定PCM信道中的噪声,确定可能的星座点阵列(92)的所述噪声的范围,以及选择星座点,使得最大负噪声 第一点保持在相邻第二点(94)的最大正噪声电平之上。 星座是离线创建并存储(96),用于在调制期间根据回波的水平进行检索。

    COMBINED DAQ/RBS COMPENSATION SYSTEM AND METHOD FOR ENHANCING THE ACCURACY OF DIGITAL DATA COMMUNICATED THROUGH A NETWORK
    23.
    发明公开
    COMBINED DAQ/RBS COMPENSATION SYSTEM AND METHOD FOR ENHANCING THE ACCURACY OF DIGITAL DATA COMMUNICATED THROUGH A NETWORK 失效
    FOR DAQ / RBS和方法用于改善的准确性通过网络的补偿方法SWITCHED DIGITAL DATA

    公开(公告)号:EP0920762A4

    公开(公告)日:2003-05-21

    申请号:EP97931428

    申请日:1997-06-25

    CPC分类号: H04B14/048 H04J3/125

    摘要: A compensation system (figure 2A) for improving the accuracy of digital signals that are communicated through a digital network (113, figure 1) by reducing loss from attenuation quantization (DAQ) and rob bit signaling (RBS). The combined DAQ/RBS compensation system can be employed within the transmitting modem (101) connected to the digital network. A first adjustment mechanism combines a DAQ compensation quantity with each segment of the digital data. Next, the word is communicated to a linear-mu-linear converter (201), which is configured to simulate a digital transmission by mu-low encoding each digital data word into a code word and then subsequently mu-low decoding each code word back into linear digital data word. The linear-mu-linear converter includes an RBS compensation system (129) that causes an RBS compensation quantity to be mathematically combined with each segment to be tainted by RBS. A second adjustment mechanism combines the reciprocal of the DAQ compensation quantity with the linear digital data from the linear-mu-linear converter. Finally, the linear digital data word is passed from the linear-mu-linear converter to a linear-mu-converter (208) for conversion into a mu-law code word and transmission to the network.

    COMBINED DAQ/RBS COMPENSATION SYSTEM AND METHOD FOR ENHANCING THE ACCURACY OF DIGITAL DATA COMMUNICATED THROUGH A NETWORK
    27.
    发明授权
    COMBINED DAQ/RBS COMPENSATION SYSTEM AND METHOD FOR ENHANCING THE ACCURACY OF DIGITAL DATA COMMUNICATED THROUGH A NETWORK 失效
    FOR DAQ / RBS和方法用于改善的准确性通过网络的补偿方法SWITCHED DIGITAL DATA

    公开(公告)号:EP0920762B1

    公开(公告)日:2006-12-13

    申请号:EP97931428.3

    申请日:1997-06-25

    IPC分类号: H04L25/49 H04B14/04 H04J3/12

    CPC分类号: H04B14/048 H04J3/125

    摘要: A compensation system (figure 2A) for improving the accuracy of digital signals that are communicated through a digital network (113, figure 1) by reducing loss from attenuation quantization (DAQ) and rob bit signaling (RBS). The combined DAQ/RBS compensation system can be employed within the transmitting modem (101) connected to the digital network. A first adjustment mechanism combines a DAQ compensation quantity with each segment of the digital data. Next, the word is communicated to a linear-mu-linear converter (201), which is configured to simulate a digital transmission by mu-low encoding each digital data word into a code word and then subsequently mu-low decoding each code word back into linear digital data word. The linear-mu-linear converter includes an RBS compensation system (129) that causes an RBS compensation quantity to be mathematically combined with each segment to be tainted by RBS. A second adjustment mechanism combines the reciprocal of the DAQ compensation quantity with the linear digital data from the linear-mu-linear converter. Finally, the linear digital data word is passed from the linear-mu-linear converter to a linear-mu-converter (208) for conversion into a mu-law code word and transmission to the network.

    Pcm upstream data transmission server
    29.
    发明公开
    Pcm upstream data transmission server 审中-公开
    变频器PCMDatenübertragungsserver

    公开(公告)号:EP1370037A2

    公开(公告)日:2003-12-10

    申请号:EP03100950.9

    申请日:2003-04-08

    IPC分类号: H04L25/03

    摘要: A method for designing an equalizer (32) and tracking performance for upstream PCM in a digital communications network is described. The invention optimizes upstream data rates for data transmissions in a network between a client modem and a server modem. During training, the upstream channel impulse response (46) is identified compensating for any robbed bit signaling. The upstream transmit equalizer (32) is computed in closed form based on the identified channel (34). An equalizer in the receiver (41) is also used to track timing and channel variations. The invention approximates bit error rate performance by looking for code violations in the trellis code decoder and tracking their frequency. The bit error rate is used to determine if the current modem parameters need to be re-designed.

    摘要翻译: 描述了用于设计数字通信网络中的上游PCM的均衡器(32)和跟踪性能的方法。 本发明优化了在客户端调制解调器和服务器调制解调器之间的网络中的数据传输的上行数据速率。 在训练期间,上行信道脉冲响应(46)被识别为补偿任何抢占位信令。 基于识别的信道(34),以封闭形式计算上行发送均衡器(32)。 接收机(41)中的均衡器也用于跟踪定时和信道变化。 本发明通过查找格码解码器中的代码违例并跟踪其频率来近似误码率性能。 误码率用于确定当前调制解调器参数是否需要重新设计。

    A Voice/Facsimile/Modem Call Discrimination Method For Voice Over Packet Networks
    30.
    发明公开
    A Voice/Facsimile/Modem Call Discrimination Method For Voice Over Packet Networks 审中-公开
    一种用于在分组网络中区分语音/传真/调制解调器呼叫用于语音方法

    公开(公告)号:EP1331803A2

    公开(公告)日:2003-07-30

    申请号:EP02102877.4

    申请日:2002-12-23

    发明人: Fruth, Frank E.

    IPC分类号: H04N1/327 H04M11/06 H04Q3/62

    摘要: A method of discriminating voice, data, and facsimile calls communicated through a voice-over-packet network. The gateway (2) is provided with software which can identify the existence of an answer signal (ANS) or a modified answer signal (ANSam) communicated between an answering modem (4) and an originating modem (1) over a packet network (5) during a voice state call. The originating gateway (2) can generate an ANS tone according to the protocols of the originating modem (1), using an originating-side gateway (2), when the existence of the ANS signal is identified by the receiving-side gateway (3). The originating gateway also generates an ANSam tone according to the protocols of the originating modem, using the originating-side gateway, when the existence of the ANSam signal is identified by the receiving-side gateway.

    摘要翻译: 识别语音,数据,并通过一个声音在分组网络传送的传真呼叫的方法。 网关(2)设置有软件,该软件能够识别的应答信号(ANS)或之间通信修正的应答信号(ANSam的)在应答调制解调器(4)的存在,并在(发端调制解调器(1)在分组网络5 )语音通话状态中。 始发端网关(2)可以在ANS音调雅丁到发端调制解调器的协议产生(1),使用在始发侧网关(2)当ANS信号的存在由接收侧网关鉴定(3 )。 由此始发端网关生成ANSam的音调gemäß到发端调制解调器的协议的速率,使用始发侧网关,当ANSam信号的存在由接收侧网关标识。