721results about How to "Improve voice quality" patented technology

A method for improving the quality of a speech signal extracted from a noisy acoustic environment is provided. In one approach, a signal separation process is associated with a voice activity detector. The voice activity detector is a two-channel detector, which enables a particularly robust and accurate detection of voice activity. When speech is detected, the voice activity detector generates a control signal. The control signal is used to activate, adjust, or control signal separation processes or post-processing operations to improve the quality of the resulting speech signal. In another approach, a signal separation process is provided as a learning stage and an output stage. The learning stage aggressively adjusts to current acoustic conditions, and passes coefficients to the output stage. The output stage adapts more slowly, and generates a speech-content signal and a noise dominant signal. When the learning stage becomes unstable, only the learning stage is reset, allowing the output stage to continue outputting a high quality speech signal.

An apparatus for providing voice conversion using temporal dynamic features includes a feature extractor and a transformation element. The feature extractor may be configured to extract dynamic feature vectors from source speech. The transformation element may be in communication with the feature extractor and configured to apply a first conversion function to a signal including the extracted dynamic feature vectors to produce converted dynamic feature vectors. The first conversion function may have been trained using at least dynamic feature data associated with training source speech and training target speech. The transformation element may be further configured to produce converted speech based on an output of applying the first conversion function.

The present invention provides a method and apparatus for text to speech conversion, and a method and apparatus for adjusting a corpus. The method for text to speech comprises: text analysis step for parsing the text to obtain descriptive prosody annotations of the text based on a TTS model generated from a first corpus; prosody parameter prediction step for predicting the prosody parameter of the text according to the result of text analysis step; speech synthesis step for synthesizing speech of said text based on said the prosody parameter of the text; wherein descriptive prosody annotations of the text include prosody structure for the text, the prosody structure of the text is adjusted according to a target speech speed for the synthesized speech. The present invention adjusts the prosody structure of the text according to the target speech speed. The synthesized speech will have improved quality.

An apparatus for providing voice conversion using temporal dynamic features includes a feature extractor and a transformation element. The feature extractor may be configured to extract dynamic feature vectors from source speech. The transformation element may be in communication with the feature extractor and configured to apply a first conversion function to a signal including the extracted dynamic feature vectors to produce converted dynamic feature vectors. The first conversion function may have been trained using at least dynamic feature data associated with training source speech and training target speech. The transformation element may be further configured to produce converted speech based on an output of applying the first conversion function.

An unlicensed wireless communication base station includes a radio frequency circuit to transmit an unlicensed wireless communication base station identification signal to a subscriber device and to receive licensed wireless communication system security information from the subscriber device responding to the unlicensed wireless communication base station identification signal. A control circuit is connected to the radio frequency circuit. Network interface circuitry is connected to the control circuit. The control circuit conveys the licensed wireless communication system security information to the network interface circuitry for delivery to a network. The network interface circuitry receives from the network an authentication command that is delivered to the control circuit. The control circuit and the radio frequency circuit coordinate unlicensed wireless communication between the subscriber device and the unlicensed wireless communication base station in response to the authentication command.

A method for improving the quality of a speech signal extracted from a noisy acoustic environment is provided. In one approach, a signal separation process is associated with a voice activity detector. The voice activity detector is a two-channel detector, which enables a particularly robust and accurate detection of voice activity. When speech is detected, the voice activity detector generates a control signal. The control signal is used to activate, adjust, or control signal separation processes or post-processing operations to improve the quality of the resulting speech signal. In another approach, a signal separation process is provided as a learning stage and an output stage. The learning stage aggressively adjusts to current acoustic conditions, and passes coefficients to the output stage. The output stage adapts more slowly, and generates a speech-content signal and a noise dominant signal. When the learning stage becomes unstable, only the learning stage is reset, allowing the output stage to continue outputting a high quality speech signal.

Speech signal classification and encoding systems and methods are disclosed herein. The signal classification is done in three steps each of them discriminating a specific signal class. First, a voice activity detector (VAD) discriminates between active and inactive speech frames. If an inactive speech frame is detected (background noise signal) then the classification chain ends and the frame is encoded with comfort noise generation (CNG). If an active speech frame is detected, the frame is subjected to a second classifier dedicated to discriminate unvoiced frames. If the classifier classifies the frame as unvoiced speech signal, the classification chain ends, and the frame is encoded using a coding method optimized for unvoiced signals. Otherwise, the speech frame is passed through to the “stable voiced” classification module. If the frame is classified as stable voiced frame, then the frame is encoded using a coding method optimized for stable voiced signals. Otherwise, the frame is likely to contain a non-stationary speech segment such as a voiced onset or rapidly evolving voiced speech signal. In this case a general-purpose speech coder is used at a high bit rate for sustaining good subjective quality.

The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

A system improves the perceptual quality of a speech signal by dampening undesired repetitive transient noises. The system includes a repetitive transient noise detector adapted to detect repetitive transient noise in a received signal. The received signal may include a harmonic and a noise spectrum. The system further includes a repetitive transient noise attenuator that substantially removes or dampens repetitive transient noises from the received signal. The method of dampening the repetitive transient noises includes modeling characteristics of repetitive transient noises; detecting characteristics in the received signal that correspond to the modeled characteristics of the repetitive transient noises; and substantially removing components of the repetitive transient noises from the received signal that correspond to some or all of the modeled characteristics of the repetitive transient noises.

A method and apparatus for reducing the audible "clicks" or "pops" which occur when an ADPCM encoding and decoding system is employed in a communications system in which communication occurs over a dispersive channel. A novel technique is employed in which ADPCM-encoded silence is substituted for error-containing frames, and post-processing is performed on decoded frames while a muting window is open.

A system and method for separating a mixture of audio signal into desired audio signals (430) (e.g., speech) and a noise sign (440) is disclosed. Microphones (310, 320) are positioned to receive the mixed audio signals, and an independent component analysis (ICA) processes (212) the sound mixture using stability constraints. The ICA process (508) uses predefined characteristics of the desired speech signal to identify and isolate a target sound signal (430). Filter coefficients are adapted with a learning rule and filter weight update dynamics are stabilized to assist convergence to a stable separated ICA signal result. The separated signals may be peripherally-processed to further reduce noise effects using post-processing (214) and pre-processing (220, 230) techniques and information. The proposed system is designed and easily adaptable for implementation on DSP units or CPUs in audio communication hardware environments.

A packet network congestion control system using a biased packet discard policy includes a plurality of end points having codecs operating in a framework, e.g. ITU-T H.323 protocol to establish a communication session. The protocol enables the codecs to negotiate codec type and associated parameters. Once a connection and session are established, compressed voice and data packets start flowing between the two end points. A control entity supplies congestion control packets periodically. The control packets provide a "heartbeat" signal to the codec at the other end of the session. Each codec receiver uses the "heartbeat" signal as an indication of network congestion. As network congestion increases, routers within the network discard excess packets to prevent network failure. The network discards all packets classified as congestion control packets whenever a flow control mechanism detects congestion or a trend toward congestion. As packets are discarded, the end points renegotiate codec type and / or parameters to realize lower bit rates.

A method and a wearable communication system for personal face-to-face and wireless communications in high noise environments are provided. A noise cancellation device (NCD) operably coupled to a wireless coupling device (WCD) includes a speech acquisition unit, an audio signal processing unit, one or more loudspeakers, and a communication module. The NCD receives voice vibrations from user speech via a contact microphone and a second microphone and converts the voice vibrations into an audio signal. The NCD processes the audio signal to remove noise signals and enhance a speech signal contained in the audio signal. A loudspeaker emits the speech signal during face-to-face communication. The NCD transmits the speech signal to a communication device via the WCD and receives an external speech signal from the communication device during wireless communication. With the NCD, the signal intelligibility and signal-to-noise ratio can be improved, for example, from −10 dB to 20 dB.

An encoder encodes digital signals representative of data by classifying the digital signals into first and second classes indicative of their influence on data quality and subjects them to error detection encoding capable of generating at least two error detection codes which respectively correspond to the first and second classes. A decoder receives the encoded digital signals classified into first and second digital signal classes, decodes the error detection codes, and generates error signals, corresponding to the respective digital signal classes, from which the quality of the received digital signals is estimated and the utility of the received digital signals is determined.

A method for handling media in a make-before-break handoff between wireless networks is provided that includes communicating with a mobile station to facilitate a session handoff between two call legs of a session. The method also includes identifying a relevant frame of one of the legs, performing a cross correlation operation, generating a delay difference estimate between the two legs, and performing a level difference estimation for the two legs.

The invention provides a method for enhancing microphone array voice based on combined inhibition. The method comprises the following steps of: structuring a microphone array for receiving external signals; analyzing the signals and obtaining time delays of different array signals relative to benchmark array signals in the microphone arrays opposite to a target voice source; respectively performing time delay compensation on digital signals corresponding to the two microphones, obtaining the compensated signals; respectively performing subband decomposition on the compensated array signals, and then forming fixed beams on each subband; meanwhile, respectively using blocking matrixes on each subband to obtain noise reference signals on each subband; and then respectively removing the noiseirrelative to the target voice form the fixed beam forming device on corresponding subband through an adaptive filtering processing algorithm, and then merging the subbands, thereby forming an initial gain signal; and meanwhile, making use of the previously compensated any two array signals to obtain a filter for inhibiting the noise signal related to the target voice through a recursive mutual power spectral density, thereby obtaining the final target voice signal through combining the initial gain signal.

The invention proposes a method and apparatus for significantly improving the quality of voice morphing and guaranteeing the similarity of converted voice. The invention sets several standard speakers in a TTS database, and selects the voices of different standard speakers for speech synthesis according to different roles, wherein the voice of the selected standard speaker is similar to the original role to a certain extent. Then the invention further performs voice morphing on the standard voice similar to the original voice to a certain extent, in order to accurately mimic the voice of the original speaker, so as to make the converted voice closer to the original voice features while guaranteeing the similarity.

The present invention helps to reduce the noise level and to enhance the quality of speech signals, in communications, computers, entertainment and other applications, where microphones and loudspeakers are involved. Additionally, the invention includes a new noise reduction and speech enhancement algorithm which is created based on the principles of human hearing mechanism. Further, the algorithm uses a new set of speech recognition parameters instead of just signal-to-noise ratio (“SNR”) as used in the prior art.

PCT No. PCT / FI95 / 00122 Sec. 371 Date Sep. 9, 1996 Sec. 102(e) Date Sep. 9, 1996 PCT Filed Mar. 3, 1995 PCT Pub. No. WO95 / 24789 PCT Pub. Date Sep. 14, 1995A mobile communication system and a call control method for the mobile communication system allowing calls to be established between mobile stations located in an area of a same base station system. Uplink transmission frames are transmitted from a first base station communicating with one of a calling and a called mobile station to a second base station communicating with another of the calling and the called mobile station via the same base station system. The mobile communication system and the call control method for the mobile communication system also allow calls to be established between mobile stations in an area of two base station systems having a direct network link therebetween. The uplink transmission frames are transmitted from the first base station communicating with one of the calling and the called mobile station to the second base station communicating with another of the calling and the called mobile station via the two base station systems having the direct network link therebetween.

A method and apparatus for concealing frame loss and an apparatus for transmitting and receiving a speech signal that are capable of reducing speech quality degradation caused by packet loss are provided. In the method, when loss of a current received frame occurs, a random excitation signal having the highest correlation with a periodic excitation signal (i.e., a pitch excitation signal) decoded from a previous frame received without loss is used as a noise excitation signal to recover an excitation signal of a current lost frame. Furthermore, a third, new attenuation constant (AS) is obtained by summing a first attenuation constant (NS) obtained based on the number of continuously lost frames and a second attenuation constant (PS) predicted in consideration of change in amplitude of previously received frames to adjust the amplitude of the recovered excitation signal for the current lost frame. Speech quality degradation caused by packet loss can be reduced for enhanced communication quality in a packet network environment with continuous frame loss.

Disclosed is a packet (frame) loss concealment method and device for a VoIP system, for reducing speech quality degradation caused by packet loss generated when transmitting speech data through a packet network. When a packet loss occur, the speech signal of the lost frame are reconstruct by combing the forward and backward prediction from the good frame before and after the lost frame Thus, speech quality of a speech coder can be improved without any extra delay in packet loss condition.

A method to eliminate discontinuities in an adaptively filtered signal includes filtering a beginning portion of a current signal frame using a past set of filter coefficients, thereby producing a first filtered frame portion. The method also includes filtering the beginning portion of the current signal frame using a current set of filter coefficients, thereby producing a second filtered frame portion. The method also includes modifying the second filtered frame portion with the first filtered frame portion so as to smooth a possible filtered signal discontinuity between the second filtered frame portion and a past filtered frame produced using the past filter coefficients.

A method and apparatus in a base station for jointly controlling sub-channel transmission power and assigned codec modes for a first and second mobile station utilizing Voice services over Adaptive Multi-user channel on One Slot (VAMOS). The base station receives signal quality information reports from the mobile stations every 480 ms using the Slow Associated Control Channel (SACCH), and receives codec mode requests from the mobile stations every 40 ms using Adaptive Multi-Rate (AMR) in-band signaling. The base station associates the requested codec modes with estimated levels of speech quality currently being experienced by the first and second mobile stations. The base station then allocates sub-channel transmission power and assigns codec modes to the first and second mobile stations based on the estimated levels of speech quality associated with the requested codec modes and the signal quality reports.

A system and method is disclosed for reconstructing high frequency components of a digital audio signal using a harmonic enhancer in a baseband integrated circuit of a receiver handset. The original spectrum of the digital audio signal is upsampled in a times two (2) upsample unit to double the size of the bandwidth. A low pass filter then removes a high frequency alias of the original spectrum. The spectrum is then modulated with a first carrier frequency and sent to a first filter bank where a low pass filter and a high pass filter shape the modulated harmonic spectrum. After gain adjustment, the modulated harmonic spectrum is added to a delayed version of the original spectrum. Additional harmonic spectra are similarly created at other carrier frequencies and added to the audio output spectra to reconstruct high frequency components of the audio signal.

A personal medical dictation system that can be easily and conveniently used to capture and preserve audio information. The system includes a specially designed portable, hand-held recording component that is of a small size, but yet is capable of storing at least one hour of actual dictation in compressed form and a cooperating dictation receiver that functions to automatically transfer the recorded data to a central processing area. The hand-held recording component can be expeditiously, mechanically, and electrically transmitted to the central processing area for transcription. The dictation receiver component also automatically recharges the batteries of the hand-held recording unit.

The present invention relates to a receiver system in a communication system supporting packet-based communicatior (e.g., an IP-network), including a receiver, speech decoder (40) and a jitter buffer (20) for handling delay variations in the reception of a speech signal consisting of packets containing frames with encoded speech. A jitter buffer controller (50) is provided for keeping information about the functional size of the jitter buffer (20) and for providing the speech decoder (40) with control information, such that the speech decoder (40), based on that information. provides a dynamic adaptation of the size of the jitter buffer (20) using the received encoded, packetized speech signal. The invention also relates to a method of adapting the functional size of the jitter buffer of a receiver system.

The present invention relates to a method with which speech is captured in a noisy environment with as high a speech quality as possible. To this end, a compact array of, for example, two single microphones is combined to form one system through signal processing methods consisting of adaptive beam formation and spectral subtraction. Through the combination with a spectral subtraction, the reference signal of the beam former is freed from speech signal components to the extent that a reference signal of the interference is formed and the beam former produces high gains.

The system includes a jitter buffer for receiving speech packets in a Voice over Internet Protocol (VoIP) system, a playback device for adjusting the playback speed of the received speed packets, and a jitter buffer manager for detecting out of sequence packets in the jitter buffer and for sending commands to the playback device to adjust playback speed based on the detection. The speech signal is played back at the nominal speed when there are no out of sequence packets. The playback speed is decreased when an out of sequence packet is detected, thereby tending to increase the jitter buffer length. When an out of sequence packet arrives, the playback speed is increased in order to restore jitter buffer length to its nominal length.

Devices, softwares and methods are provided for redundantly encoding a data stream into frames for network transmission as packets. A main encoder encodes a data stream into main frames, while a redundant encoder encodes it into redundant frames. The redundant frames have a redundant-coding delay from the main frames that is adjustable to accommodate many different sets of network conditions.