In lossy based lossless coding a PCM audio signal passes through a lossy encoder (101) to a lossy decoder (102). The lossy encoder provides a lossy bit stream (111). The lossy decoder also provides side information (115) that is used to control (105) the coefficients (118) of a prediction filter (106) that de-correlates the difference signal (104) between the PCM signal and the lossy decoder output. The de-correlated difference signal is lossless encoded (108), providing an extension bit stream (121). Instead of, or in addition to, de-correlating in the time domain, a de-correlation in the frequency domain using spectral whitening can be performed. The lossy encoded bit stream together with the lossless encoded extension bit stream form a lossless encoded bitstream. The invention facilitates enhancing a lossy perceptual audio encoding/decoding by an extension that enables mathematically exact reproduction of the original waveform, and provides additional data for reconstructing at decoder site an intermediate-quality audio signal. The lossless extension can be used to extend the widely used mp3 encoding/decoding to lossless encoding/decoding and superior quality mp3 encoding/decoding.

Method and apparatus for lossless encoding of a source signal, using a lossy encoded data stream and a lossless extension data stream

Provided is an audio decoding device performing frame loss compensation capable of obtaining a decoded audio which is natural for ears with little noise. The audio decoding device includes: a non-cyclic pulse waveform detection unit (19) for detecting a non-cyclic pulse waveform section in a n-1-th frame which is repeatedly used with a pitch cycle in the n-th frame upon compensation of loss of the n-th frame; a non-cyclic pulse waveform suppression unit (17) for suppressing a non-cyclic pulse waveform by replacing an audio source signal existing in the non-cyclic pulse waveform section in the n-1-th frame by a noise signal; and a synthesis filter (20) for using a linear prediction coefficient decoded by an LPC decoding unit (11) to perform synthesis by a synthesis filter by using the audio source signal of the n-1-th frame from the non-cyclic pulse waveform suppression unit (17) as a drive audio source, thereby obtaining the decoded audio signal of the n-th frame.

Audio decoding device and audio decoding method

Methods and apparatus for coordinating audio data processing and network communication processing in a communication device. In an exemplary method lower and upper threshold values for use by a network communication processing circuit are set, the lower and upper threshold values defining a window of timing offsets relative to each of a series of periodic network communications frame boundaries. A series of encoded audio data frames are sent to the network communication processing circuit for transmission over the network communications link. The delivery of encoded audio data to the network communication processing circuit outside of the corresponding time window defined by the threshold values will trigger an event report. This event report is received from the network communication processing circuit by the audio data processing circuit, and, in response, timing is adjusted for the sending of one or more of the encoded audio data frames.

Minimizing Speech Delay in Communication Devices

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.

Frame loss concealment method and device for VoIP system

A method of receiving at a terminal a first signal transmitted via a communication network, said method comprising the steps of; receiving at the terminal the first signal comprising a plurality of data elements; analysing characteristics of the first signal; receiving from a user of the terminal a second signal to be transmitted from the terminal; analysing characteristics of the second signal to detect audio activity in the second signal; and applying a delay between receiving at the terminal and outputting from the terminal at least one of said plurality of data elements; and adjusting the delay based on the analysed characteristics of the first signal and on the detection of audio activity in the second signal.

Method of transmitting data in a communication system

Provided are a speech restoration system and method for concealing packet losses. The system includes a demultiplexer that demultiplexes an input bit stream and divides the input bit stream into several packets; a packet loss concealing unit that produces and outputs a linear spectrum pair (LSP) coefficient representing the vocal tract of voice and an excitation signal corresponding to a lost frame, when a packet loss occurs; and a speech restoring unit that synthesizes voice using the packets input from the demultiplexer, outputs the result as restored voice, and synthesizes voice corresponding to a lost packet using the LSP coefficient and the excitation signal input from the packet loss concealing unit and outputs the result as restored voice when the lost packet is detected, wherein the packet loss concealing unit repeats linear prediction coefficients (LPCs) of a last-received valid frame, produces a first excitation signal for the lost frame using a time scale modification (TSM) method, when the lost frame is voiceless, and produces a second excitation signal by re-estimating a gain parameter based on the first excitation signal, when the lost frame is voiced.

Speech restoration system and method for concealing packet losses

The present invention relates to a transmitter and a receiver for speech coding and decoding by using an additional bit allocation method. The transmitter and the receiver according to the present invention realize a voice communication service of high quality by using additional bits permitted in system requirements while using a conventional speech coder as it is. In addition, the transmitter and the receiver according to the present invention have an advantage in that they enable insertion of additional quantization blocks while not changing the structure of the conventional standard speech coder, since they allocate additional bits by applying a multi-stage quantization procedure not in a speech signal domain but in a parameter domain.

Transmitter and receiver for speech coding and decoding by using additional bit allocation method

The system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, a transmitter analysis filter for receiving the digital speech signal and dividing it into a baseband signal and an enhancement residual band signal, a standard baseband encoder for accepting the baseband signal and coding it using an ITU-T encoder, an additional baseband encoder for reducing standard coding distortion in the baseband signal, an enhancement residual band encoder for coding a signal obtained by removing the coded baseband signal from the original digital speech signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

System and method for transmitting and receiving wideband speech signals

A system for transmitting and receiving a wideband speech signal includes an A/D converter for receiving an analog speech signal to convert it into a digital speech signal, encoding a difference between a baseband speech signal and a standard baseband between a synthesized standard baseband signal, and an IP network interface for multiplexing the coded standard and additional baseband signals and enhancement residual band signal.

System and method for transmitting and receiving wideband speech signals with a synthesized signal

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Speech restoration system and method for concealing packet losses

Transmitter and receiver for speech coding and decoding by using additional bit allocation method

System and method for transmitting and receiving wideband speech signals

System and method for transmitting and receiving wideband speech signals with a synthesized signal