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1. 发明授权

US06757654B1 Forward error correction in speech coding 有权
标题翻译：语音编码中的前向纠错
公开(公告)号：US06757654B1
公开(公告)日：2004-06-29
申请号：US09569312
申请日：2000-05-11
申请人： Magnus Westerlund , Anders Nohlgren , Jonas Svedberg , Anders Uvliden , Jim Sundqvist
发明人： Magnus Westerlund , Anders Nohlgren , Jonas Svedberg , Anders Uvliden , Jim Sundqvist
IPC分类号： G10L1302
CPC分类号： G10L19/005
摘要： An improved forward error correction (FEC) technique for coding speech data provides an encoder module which primary-encodes an input speech signal using a primary synthesis model to produce primary-encoded data, and redundant-encodes the input speech signal using a redundant synthesis model to produce redundant-encoded data. A packetizer combines the primary-encoded data and the redundant-encoded data into a series of packets and transmits the packets over a packet-based network, such as an Internet Protocol (IP) network. A decoding module primary-decodes the packets using the primary synthesis model, and redundant-decodes the packets using the redundant synthesis model. The technique provides interaction between the primary synthesis model and the redundant synthesis model during and after decoding to improve the quality of a synthesized output speech signal. Such “interaction,” for instance, may take the form of updating states in one model using the other model.
摘要翻译：用于编码语音数据的改进的前向纠错（FEC）技术提供了一种编码器模块，其使用主要合成模型对输入语音信号进行一次编码以产生初始编码数据，并且使用冗余合成模型对输入语音信号进行冗余编码以产生冗余编码数据。分组器将主编码数据和冗余编码数据组合成一系列分组，并通过诸如因特网协议（IP）网络的基于分组的网络传送分组。解码模块使用主合成模型对分组进行主要解码，并使用冗余合成模型对分组进行冗余解码。该技术在解码期间和之后提供主要合成模型和冗余合成模型之间的交互以提高合成输出语音信号的质量。例如，这种“交互”可以采用其他模型在一个模型中更新状态的形式。

2. 发明授权

US06230130B1 Scalable mixing for speech streaming 失效
标题翻译：语音流可扩展混音
公开(公告)号：US06230130B1
公开(公告)日：2001-05-08
申请号：US09025375
申请日：1998-05-18
申请人： Paulo M. Castello da Costa , Nermeen Ismail , Ross P. Morley , Atul N. Sinha
发明人： Paulo M. Castello da Costa , Nermeen Ismail , Ross P. Morley , Atul N. Sinha
IPC分类号： G10L1302
CPC分类号： H04N21/4341 , G10L19/04 , G10L19/167 , H04M3/568 , H04N21/4394 , H04N21/8106
摘要： A speech processing system receives multiple streams of speech frames. The system selects among concurrent ones of the frames a subset of those frames that are the most relevant, based on pre-assigned stream priorities and energy content of the frames. The selected frames are then decoded and rendered. The resulting signals are mixed. This architecture provides bandwidth scalability and/or processing power scalability.
摘要翻译：语音处理系统接收多个语音帧流。该系统基于预分配的流优先级和帧的能量内容，选择并发的那些帧中那些是最相关的帧的子集。然后对所选择的帧进行解码和呈现。产生的信号混合。该架构提供带宽可扩展性和/或处理能力可扩展性。

3. 发明授权

US06208958B1 Pitch determination apparatus and method using spectro-temporal autocorrelation 有权
标题翻译：使用频谱自相关的音调确定装置和方法
公开(公告)号：US06208958B1
公开(公告)日：2001-03-27
申请号：US09226115
申请日：1999-01-07
申请人： Yong-duk Cho , Moo-Young Kim
发明人： Yong-duk Cho , Moo-Young Kim
IPC分类号： G10L1302
CPC分类号： G10L25/90 , G10L25/06
摘要： A pitch determination apparatus and method using spectro-temporal autocorrelation to prevent pitch determination errors are provided. The pitch determination apparatus using spectro-temporal autocorrelation includes a formant bandwidth extension unit for extending a formant bandwidth to reduce the influence of the first formant with respect to an input voice, a temporal autocorrelation calculation unit for calculating an autocorrelation value of a time axial voice within a candidate pitch range with respect to a time axial speech signal output from the formant bandwidth extension unit, a spectral autocorrelation calculation unit for transforming the time axial speech signal output from the formant bandwidth extension unit into a frequency axial signal, and calculating an autocorrelation value between frequency axis amplitude spectrums within the candidate pitch range, an autocorrelation value synthesis unit for summing the autocorrelation values obtained by the temporal and spectral autocorrelation calculation units and obtaining a spectro-temporal autocorrelation value, and a pitch determination unit for determining a pitch having a maximum spectro-temporal autocorrelation value as a final pitch. According to this apparatus, pitch determination errors are reduced by determining a pitch using the temporal and spectral autocorrelation values, thus improving the quality of speech communication.
摘要翻译：提供了使用频谱自相关以防止音调确定误差的音调确定装置和方法。使用频谱自相关的音调确定装置包括：共振峰带宽扩展单元，用于扩展共振峰带宽以减少第一共振峰相对于输入声音的影响;时间自相关计算单元，用于计算时间轴向声音的自相关值在从共振峰带宽扩展单元输出的时间轴向语音信号的候选音调范围内，频谱自相关计算单元，用于将从共振峰带宽扩展单元输出的时间轴语音信号变换为频率轴向信号，并计算自相关在候选音调范围内的频率轴振幅谱之间的值;自相关值合成单元，用于对由时间和频谱自相关计算单元获得的自相关值求和并获得谱时间自相关值;以及音调确定单元，用于确定具有作为最终间距的最大频谱自相关值的音调。根据该装置，通过使用时间和频谱自相关值确定音调来减小音调确定误差，从而提高语音通信的质量。

4. 发明授权

US06708153B2 Voice site personality setting 失效
公开(公告)号：US06708153B2
公开(公告)日：2004-03-16
申请号：US10189156
申请日：2002-07-02
申请人： Paul St. John Brittan , Robert Francis Squibbs
发明人： Paul St. John Brittan , Robert Francis Squibbs
IPC分类号： G10L1302
CPC分类号： G10L13/08 , G10L15/30
摘要： A method is provided of setting the voice personality of a voice service site. A user browsing a voice web visits a voice site where the voice output of the site is presented using a set of voice personality characterisers with which the user is particularly comfortable. The user, in subsequently transferring to another voice service site, opts to have the voice personality that was embodied in the set of voice personality characterisers used by the site being left, transfer with the user to the new site. This transfer will typically be subject to permissions set by both the site being left and the site about to be visited.

5. 发明授权

US06574600B1 Audio financial data system 失效
标题翻译：音频财务数据系统
公开(公告)号：US06574600B1
公开(公告)日：2003-06-03
申请号：US09483286
申请日：2000-01-14
申请人： Bradley S. Fishman , Wade J. Vagle
发明人： Bradley S. Fishman , Wade J. Vagle
IPC分类号： G10L1302
CPC分类号： G06Q40/06 , G06Q40/00 , G10L13/02 , G10L25/48
摘要： A financial data system is disclosed that receives real-time data, uses a set of pre-determined rules to prioritize the data and provide a priority value, and then delivers the highest priority data by way of multiple audio channels. A key aspect of the invention is the use of data manipulation according to the priority value to adjust delivery volume, provide selective vocalization compression, add additional audio channels, or to override an existing comment when required. As a result of the invention, a significant amount of information may be aurally delivered to a user including properties of events as they change in response to changing financial conditions.
摘要翻译：公开了一种接收实时数据的财务数据系统，使用一组预先确定的规则来优先处理数据并提供优先级值，然后通过多个音频信道传送最高优先级的数据。本发明的一个关键方面是使用根据优先级值的数据操纵来调整传送量，提供选择性声音压缩，添加附加音频通道，或者在需要时覆盖现有注释。作为本发明的结果，当响应于不断变化的财务状况变化时，大量的信息可以被听觉地传递给用户，包括事件的属性。

6. 发明授权

US06304845B1 Method of transmitting voice data 失效
标题翻译：发送语音数据的方法
公开(公告)号：US06304845B1
公开(公告)日：2001-10-16
申请号：US09632338
申请日：2000-08-03
申请人： Klaus Hünlich , Wolfgang Fraas
发明人： Klaus Hünlich , Wolfgang Fraas
IPC分类号： G10L1302
CPC分类号： G10L15/07 , G10L15/16 , G10L19/0018
摘要： In a transmission of voice data, the stream of voice data is first decomposed into phonemes. For each phoneme a code symbol which is assigned to that specific phoneme in a selectable voice-specific and/or speaker-specific phoneme catalog is transmitted to a voice synthesizer at the transmission destination. The amount of data which has to be transmitted is generally greatly reduced. The decomposition of the stream of voice data into phonemes is carried out by a neural network which is trained to detect the phonemes stored in the selected voice-specific and/or speaker-specific phoneme catalog. The voice synthesizer reconverts the stream of received code symbols into a sequence of phonemes and outputs it.
摘要翻译：在语音数据的传输中，语音数据流首先被分解为音素。对于每个音素，在可选择的特定语音和/或扬声器特定音素目录中分配给该特定音素的代码符号被发送到发送目的地的语音合成器。必须传输的数据量通常大大降低。将语音数据流分解成音素由神经网络进行，该神经网络被训练以检测存储在所选择的特定语音和/或扬声器特定音素目录中的音素。语音合成器将接收到的代码符号流转换为音素序列并输出。

7. 发明授权

US06675144B1 Audio coding systems and methods 有权
标题翻译：音频编码系统和方法
公开(公告)号：US06675144B1
公开(公告)日：2004-01-06
申请号：US09423758
申请日：2000-02-16
申请人： Roger Cecil Ferry Tucker , Carl William Seymour , Anthony John Robinson
发明人： Roger Cecil Ferry Tucker , Carl William Seymour , Anthony John Robinson
IPC分类号： G10L1302
CPC分类号： G10L21/038 , G10L19/005 , G10L19/087
摘要： An audio signal is decomposed into lower and upper sub-band and at least the noise component of the sub-band is encoded. At the decoder the audio signal is synthesised by a decoding means which utilises a synthesised noise excitation signal and a filter to reproduce the noise component in the upper sub-band.
摘要翻译：音频信号被分解为下部和上部子带，并且至少对该子带的噪声分量进行编码。在解码器处，音频信号由利用合成噪声激励信号和滤波器的解码装置合成以再现上部子带中的噪声分量。

8. 发明授权

US06317713B1 Speech synthesis based on cricothyroid and cricoid modeling 有权
标题翻译：基于甲状腺和环状模型的语音合成
公开(公告)号：US06317713B1
公开(公告)日：2001-11-13
申请号：US09155156
申请日：1999-01-06
申请人： Seiichi Tenpaku
发明人： Seiichi Tenpaku
IPC分类号： G10L1302
CPC分类号： G10L13/04 , G10L13/08 , G10L13/10
摘要： Sound generating parameters are used for outputting fundamental frequency and a command regarding prosody, and a sound source generator. The sound generation device further includes use of an accent command and a descent command for calculating fundamental frequency and incorporates a rhythm command, which is representable by a sine wave. The device also uses character string analysis for analyzing a character string and generating a command concerning phoneme and prosody, a calculating element for outputting fundamental frequency as sound generation parameters, which depends on prosody, a sound source generator, and an articulator that depends on a phoneme command.
摘要翻译：声音产生参数用于输出基频和有关韵律的命令，以及声源发生器。声音产生装置还包括使用重音命令和下降命令来计算基本频率，并且包括可由正弦波表示的节奏指令。该装置还使用字符串分析来分析字符串并产生关于音素和韵律的命令，用于输出基本频率作为声音产生参数的计算元件，其依赖于韵律，声源发生器和依赖于音素命令。

9. 发明授权

US06278974B1 High resolution speech synthesizer without interpolation circuit 失效
标题翻译：高分辨率语音合成器无插补电路
公开(公告)号：US06278974B1
公开(公告)日：2001-08-21
申请号：US08976155
申请日：1997-11-21
申请人： James J. Y. Lin
发明人： James J. Y. Lin
IPC分类号： G10L1302
CPC分类号： G10L13/047
摘要： The present invention is related to a speech synthesizer which includes a sampled signal storing device storing therein a sampled signal and outputting the sampled signal in response to an input signal, and a speech signal synthesizing circuit electrically connected to the sampled signal storing device, receiving an operation signal, having the sampled signal outputted by the sampled signal storing device be repeatedly operated in response to the operation signal, and then outputting a speech synthesized signal, wherein a frequency of the operation signal is higher than that of the input signal to allow the sampled signal to be repeatedly operated during a single cycle of the input signal. The present invention proceeds a plurality of times of operation for each entry of data in the storing device so that the synthesizing performance of the present synthesizer can be improved without increasing the storage amount of the sampled signals.
摘要翻译：本发明涉及一种语音合成器，其包括：采样信号存储装置，其中存储采样信号并响应于输入信号输出采样信号;以及语音信号合成电路，电连接到采样信号存储装置，具有由取样信号存储装置输出的取样信号的操作信号响应于操作信号反复操作，然后输出语音合成信号，其中操作信号的频率高于输入信号的频率，以允许采样信号在输入信号的单个周期期间重复操作。本发明对于存储装置中的数据的每个条目进行多次操作，使得可以在不增加采样信号的存储量的情况下改善本合成器的合成性能。

10. 发明授权

US06208969B1 Electronic data processing apparatus and method for sound synthesis using transfer functions of sound samples 失效
标题翻译：电子数据处理装置和声音合成方法，使用声音样本的传递函数
公开(公告)号：US06208969B1
公开(公告)日：2001-03-27
申请号：US09122520
申请日：1998-07-24
申请人： Steven DeArmond Curtin
发明人： Steven DeArmond Curtin
IPC分类号： G10L1302
CPC分类号： G10L13/04 , G10H1/125 , G10H7/00 , G10H2250/191 , G10H2250/251 , G10H2250/625
摘要： A method and an electronic data processing apparatus for wave synthesis that retains the true qualities of naturally occurring sounds, such as those of musical instruments, speech, or other sounds. Transfer functions representative of recorded sound samples are pre-calculated and stored for use in an interpolative process to generate a transfer function representative of the sound to be synthesized. The preferred transfer functions are Chebyshev polynomial-based transfer functions, which assure a highly predictable harmonic content of synthesized sound. Output sound generation is driven by time domain signals produced by reconversion of a sequence of interpolated transfer functions. Non-harmonic sounds are synthesized using multiple frequency inputs to the reconverting (waveshaping) stage, or by parallel waveshaping stages. Speech sibilants and noise envelopes of instruments are synthesized by the input of noise into the waveshaping stage by modulation of a sinusoid with band-limited noise.
摘要翻译：一种用于波合成的方法和电子数据处理装置，其保持诸如乐器，语音或其他声音的天然发音的真实质量。表示记录声音样本的传递函数被预先计算并存储以用于内插处理，以产生表示要合成的声音的传递函数。优选的传递函数是基于切比雪夫多项式的传递函数，确保合成声音的高度可预测的谐波含量。输出声音产生由通过内插传递函数序列的重新转换而产生的时域信号驱动。使用多个频率输入到再转换（波形整形）阶段或通过平行波形成形阶段来合成非谐波声音。通过调制具有带限噪声的正弦波，将噪声的声音信号和噪声包络合成为波形成形阶段。

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