An aspect of the present invention relates to a method for low-frequency emphasizing the spectrum of a sound signal transformed in a frequency domain and comprising transform coefficients grouped in a number of blocks, in which a maximum energy for one block is calculated and a position index of the block with maximum energy is determined, a factor is calculated for each block having a position index smaller than the position index of the block with maximum energy, and for each block a gain is determined from the factor and is applied to the transform coefficients of the block.

Methods and devices for low-frequency emphasis during audio compression based on acelp/tcx

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 (180) is associated with a voice activity detector (185). The voice activity detector (185) is a two-channel (178,182) detector, which enables a particularly robust and accurate detection of voice activity. When a speech is detected, the voice activity detector generates a control signal (411). The control signal (411) is used to activate, adjust, or control signal separation processes or post -processing operations (195) to improve the quality of the resulting speech signal. In another approach, a signal separation process (180) is provided as a learning stage (752) and an output stage (756). The learning stage (752) aggressively adjus to current acoustic conditions and passes coefficients to the output stage (756). The output stage (756) adapts more slowly and generates a speech-content signal (181,770) and a noise dominant signal (407,773). When the learning stage (752) becomes unstable only the learning stage (752) is reset, allowing the output stage (756) to continue outputting a high quality speech signal.

Robust separation of speech signals in a noisy environment

A headset (12) is constructed to generate an acoustically distinct speech signal in a noisy acoustic environment. The headset positions a pair of spaced-apart microphones (32-33) near a user's mouth. The microphones each receive the user’s speech, and also receive acoustic environmental noise (267). The microphone signals, which have both a noise and information component, are received into a separation process (355). The separation process generates a speech signal (356) that has a substantial reduced noise component. The speech signal is then processed for transmission (368). In one example, the transmission process includes sending the speech signal (370) to a local control module (14) using a Bluetooth radio (27).

Headset for separation of speech signals in a noisy environment

An E-9-1-1 voice-over-IP (VoIP) solution is provided wherein a 911 call from a wireless VoIP device is routed directly to the correct Public Safety Answer Point (PSAP) via dedicated trunks, together with correct location information and call-back number. VoIP gateways are implemented locally, at least one per LATA, and accept VoIP packetized data inbound, and convert it to standard wireline voice calls. Calls are routed to an IP address at the VoIP gateway, which then egresses the call to a voice port at a selective router. Dedicated voice trunks (CAMA, SS7, FG-D) are installed between each local VoIP gateway and appropriate selective routers. An Automatic Location Identification (ALI) database is provisioned with ESRKs dedicated for VoIP use. TCP/IP circuits may be established between some or all of the various local VoIP gateways.

Enhanced E911 location information using voice over internet protocol (VoIP)

The speech encoding apparatus (220) has a first layer encoding section (2201), a first layer decoding section (2202), a delay section (2203), a subtracting section (104), a frequency domain transforming section (101), a second layer encoding section (105) and a multiplexing section (106).

Encoding device, decoding device, and method thereof

The invention relates to a method for managing a packet switched, centralized conference call between a plurality of terminals (13). In order to enable an enhancement of the user comfort, it is proposed that the method comprises at a conference call server (12) receiving data packets from all terminals (13). Based on these data packets, then at least one terminal (13) currently providing voice data is determined. In a next step, the data received in the data packets is mixed, and the mixed data is inserted into new data packets together with at least one identifier associated to one of the terminals (13) which were determined to provide voice data, such that the at least one identifier can be distinguished from any other information in the data packets. Finally, the new data packets are transmitted to terminals (13) participating in the conference call. The invention relates equally to a corresponding server and to a corresponding terminal.

Managing a packet switched conference call

An error concealment method for multi-channel digital audio involves receiving an audio signal having audio data forming a first audio channel and a second audio channel included therein, wherein the first and second audio channels are correlated with each other in a manner so that a spatial sensation is typically perceived when listened to by a user. Erroneous first-channel data is detected in the first audio channel, and second-channel data is obtained from the second audio channel. The erroneous first-channel data of the first audio channel is corrected by using the second-channel data. Upon detection of the erroneous first-channel data, a spatially perceivable inter-channel relation between the first and second audio channels is determined, and the determined inter-channel relation is used when correcting the erroneous first-channel data of the first audio channel so as to preserve the spatial sensation perceived by the user.

Enhanced error concealment for spatial audio

A method of speech coding a sampled speech signal using long term prediction (LTP). A LTP pitch-lag parameter is determined for each frame of the speech signal by first determining the autocorrelation function for the frame within the signal, between predefined maximum and minimum delays. The autocorrelation function is then weighted to emphasize the function for delays in the neighborhood of the pitch-lag parameter determined for the most recent voiced frame. The maximum value for the weighted autocorrelation function is then found and identified as the pitch-lag parameter for the frame.

Pitch-lag estimation in speech coding

The invention relates to an encoder (1) comprising an input (1.2) for inputting frames of an audio signal in a frequency band, an analysis filter (1.3) for dividing the frequency band into at least a lower frequency band and a higher frequency band, a first encoding block (1.4.1) for encoding the audio signals of the lower frequency band, a second encoding block (1.4.2) for encoding the audio signals of the higher frequency band, and a mode selector for selecting operating mode for the encoder among at least a first mode and a second mode. In the first mode signals only on the lower frequency band are encoded, and in the second mode signals on both the lower and higher frequency band are encoded. The encoder (1) further comprises a scaler to control the second encoding block (1.4.2) to gradually change the encoding properties of the second encoding block (1.4.2) in connection with a change in the operating mode of the encoder. The invention also relates to a device, a decoder, a method, a module, a computer program product, and a signal.

Coding of audio signals

A device is disclosed that makes packetized and encoded speech data audible to a listener, as is a method for operating the device. The device includes a unit for generating a synchronization request for reducing an amount of synchronization delay, and further includes a speech decoder that is responsive to the synchronization delay adjustment request for executing a time-warping operation for one of lengthening or shortening a duration of a speech frame. In one embodiment the speech decoder comprises a code excited linear prediction (CELP) speech decoder, and the CELP decoder time-warping operation is applied to a reconstructed excitation signal u(k) to derive a time-warped reconstructed signal u w (k). The time-warped reconstructed signal u w (k) is input to a Linear Predictor (LP) synthesis filter to derive a CELP decoder time-warped output signal y ^ w (k). In another embodiment the speech decoder comprises a parametric speech decoder, and where an adaptation of the frame length N in the parametric speech decoder results in the use of a modified frame length N w .

Ari Lakaniemi

Papers

Managing a packet switched conference call

Enhanced error concealment for spatial audio

Pitch-lag estimation in speech coding

Coding of audio signals

Method and apparatus for reducing synchronization delay in packet switched voice terminals using speech decoder modification