103 results about "Packet loss concealment" patented technology

A technique for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system is provided. In accordance with the technique, one or more received frames in the series of frames are decoded to generate a full-band output audio signal, wherein the full-band output audio signal comprises a combination of at least a first sub-band decoded audio signal and a second sub-band decoded audio signal. The full-band output audio signal corresponding to the one or more received frames is stored. Then, a full-band output audio signal corresponding to the lost frame is synthesized, wherein synthesizing the full-band output audio signal corresponding to the lost frame comprises performing waveform extrapolation based on the stored full-band output audio signal corresponding to the one or more received frames.

A method of significantly reducing error propagation due to voice packet loss, while still greatly profiting from long-term pitch prediction, is achieved by adaptively limiting the maximum value of the pitch gain for the first pitch cycle within one frame. A speech coding system for encoding a speech signal, wherein said a plurality of speech frames are classified into said a plurality of classes depending on if the first pitch cycle is included in one subframe or several subframes. The pitch gain is set to a value significantly smaller than 1 for the subframes covering first pitch cycle; wherein the pitch gain reduction is compensated by increasing the coded excitation codebook size or adding one more stage of excitation for the subframes covering the first pitch cycle.

An error concealment method and device for recovering lost data in the AAC bitstream in the compressed domain. The bitstream are partitioned into frames each having a plurality of data parts including the header/global gain, scale factors and QMDCT coefficients. The data parts are stored in a plurality of buffers, so that if one or more data parts of a current frame is corrupted or lost, the corresponding data part in the neighboring frames is used to conceal the errors in the current frame.

A method to improve packet loss concealment for generation of a synthetic speech signal in a algebraic code excited linear prediction decoder for a voice over packet network. One method improves features for coding gains in the decoder and for post-filtering of the signals. An alternative method uses a classification method for the signal based on the bitstream in the decoder.

In a voice packet communication system, a voice packet loss concealment device compensates for the deterioration of voice quality due to voice packet loss. In the device, a detecting section detects a loss of a voice packet and outputting information; an estimating section estimates the voice characteristics of the lost segment using a pre-loss voice packet received before the lost segment or a post-loss voice packet received after the lost segment; a pitch signal generating section generates a pitch signal having the voice characteristics; and a lost packet generating section outputs the pitch signal generated by the pitch signal generating section, with the voice characteristics estimated by the estimating section, which allows abnormal noise and feeling of mute, subjective deterioration of naturalness and continuity to be improved, and the voice packet loss concealment to be further improved.

Various techniques are disclosed for improving packet loss concealment to reduce artifacts by using audio character measures of the audio signal. These techniques include attenuation to a noise fill instead of attenuation to silence, varying how long to wait before attenuating the extrapolation, varying the rate of attenuation of the extrapolation, attenuating periodic extrapolation at a different rate than non-periodic extrapolation, and performing period extrapolation on successively longer fill data based on the audio character measures, adjusting weighting between periodic and non-periodic extrapolation based on the audio character measures, and adjusting weighting between periodic extrapolation and non-periodic extrapolation non-linearly.

A packet loss concealment method and system is described that attempts to reduce or eliminate destructive interference that can occur when an extrapolated waveform representing a lost segment of a speech or audio signal is merged with a good segment after a packet loss. This is achieved by guiding a waveform extrapolation that is performed to replace the bad segment using a waveform available in the first good segment or segments after the packet loss. In another aspect of the invention, a selection is made between a packet loss concealment method that performs the aforementioned guided waveform extrapolation and one that does not. The selection may be made responsive to determining whether the first good segment or segments after the packet loss are available and also to whether a segment preceding the lost segment and the first good segment following the lost segment are deemed voiced.

A method to reduce memory requirements for a packet loss concealment algorithm in the event of packet loss in a receiver of pulse code modulated voice signals. Packet losses are concealed by using the spectral analysis filter memory to smooth a signal gap and by using a technique for determining a maximum repeatable waveform range instead of using the pitch period to reproduce lost packets. The invention uses fewer processing resources and results in improved performance compared to a packet loss concealment algorithm under G.711 Appendix I standards.

Real-time packet-based audio communications over packet-based networks frequently results in the loss of one or more packets during any given communication session. The real-time nature of such communications precludes retransmission of lost packets due to the unacceptable delays that would result. Consequently, packet loss concealment methods are employed to “hide” lost packets from the listener. Unfortunately, conventional loss concealment methods, such as packet repetition or stretch/overlap methods, do not fully exploit information available from partially received samples. Therefore, when a single frame of N coefficients is lost, 2N samples are only partially reconstructed, thereby degrading the reconstructed signal. To address this problem, an optimized packet loss concealment solution is identified for particular lost packets by solving an underdetermined system of linear equations representing partially received samples while minimizing a computed error based on a model of the signal obtained from neighboring blocks or frames received by the decoder.

The embodiment of the invention discloses a method and a device for realizing packet loss concealment. Correlation between recovered lost frame data and data after frame loss is enhanced by the technical scheme for recovering lost frames by combining the data before the frame loss with the data after the frame loss so as to improve the continuity of phases between the recovered lost frame data and the data after the frame loss and improve the quality of voice data.

A technique is described for concealing the effect of a lost frame in a series of frames representing an encoded audio signal in a sub-band predictive coding system. In accordance with the technique, a first synthesized sub-band audio signal is synthesized, wherein synthesizing the first synthesized sub-band audio signal comprises performing waveform extrapolation based on a stored first sub-band decoded audio signal. A second synthesized sub-band audio signal is also synthesized, wherein synthesizing the second synthesized sub-band audio signal comprises performing waveform extrapolation based on the stored second sub-band decoded audio signal. The first synthesized sub-band audio signal and the second synthesized sub-band audio signal are combined to generate a synthesized full-band output audio signal corresponding to a lost frame.

The present document relates to audio signal processing in general, and to the concealment of artifacts that result from loss of audio packets during audio transmission over a packet-switched network, in particular. A method (200) for concealing one or more consecutive lost packets is described. A lost packet is a packet which is deemed to be lost transform-based audio decoder. Each of the one or more lost packets comprises a set of transform coefficients. A set of transform coefficients is used by the transform-based audio decoder to generate a corresponding frame of a time domain audio signal. The method (200) comprises determining (205) for a current lost packet of the one or more lost packets a number of preceding lost packets from the one or more lost packets; wherein the determined number is referred to as a loss position. Furthermore, the method comprises determining a packet loss concealment, referred to as PLC, scheme based on the loss position of the current packet; and determining (204, 207, 208) an estimate of a current frame of the audio signal using the determined PLC scheme (204, 207, 208); wherein the current frame corresponds to the current lost packet.

A speech coding method of significantly reducing error propagation due to voice packet loss, while still greatly profiting from a pitch prediction or Long-Term Prediction (LTP), is achieved by limiting or reducing a pitch gain only for the first subframe or the first two subframes within a speech frame. The method is used for a voiced speech class; a pitch cycle length is compared to a subframe size to decide to reduce the pitch gain for the first subframe or the first two subframes within the frame. Speech coding quality loss due to the pitch gain reduction is compensated by increasing a bit rate of a second excitation component or adding one more stage of excitation component only for the first subframe or the first two subframes within the speech frame.

The invention discloses a network audio packet loss concealment method and device, relating to the field of audio transmission. The method comprises the steps of decoding an audio stream subjected to packet loss, reversing audio signals obtained through decoding, then, carrying out packet loss compensation on the audio signals and the reversed audio signals respectively, reversing the compensated reversed audio signals again, and finally, carrying out magnitude average on the compensated audio signals of two channels, so as to obtain output signals. By the bidirectional-prediction error concealment technology, the quality of audio can be further improved, and particularly, a relatively clear voice effect can be obtained in the network environment that the packet loss rate is relatively high.

The embodiment of the invention discloses an audio encoding method, an audio decoding method, a related device and a communication system. The audio encoding method comprises the steps of receiving packet loss rate fed back by an audio decoder according to acquired code streams, switching encoding capacity to a multiple description state corresponding to the packet loss rate according to the packet loss rate fed back and encoding audio data by the multiple description state. The audio decoding method comprises the steps of acquiring multiple description packets sent by the audio encoder, increasing the value of a counter of the multiple description packet which contains multiple description quantity and packet description number by 1, acquiring the numerical value of the packet loss packet, and calling packet loss concealment when determining the numerical value of the counter of the multiple description packet or when the packet description number is not smaller than the multiple description quantity and the numerical number of the packet loss counter is not smaller than the multiple description quantity. The embodiment of the invention adjusts the packet loss resisting capacity of encoding code streams according to actual condition and reduces bandwidth occupied by invalid redundant information, thereby acquiring better output code stream quality.

A modem architecture that supports the application of joint source channel decoding (JSCD). The modem architecture includes two channel decoders, one of which is modified to provide improved signal quality. The modem architecture further includes transparent network layers that enable the passage of data from one layer to another layer. For example, the modem architecture enables the passage soft bits, when available, from a physical layer to an application layer. The soft bits may be utilized for JSCD, packet loss concealment, or other applications. The modem architecture enables encryption and decryption of data to incorporate extrinsic information in operating JSCD.

Disclosed are a packet loss concealment method and apparatus a using a generative adversarial network. A method for packet loss concealment in voice communication may include training a classification model based on a generative adversarial network (GAN) with respect to a voice signal including a plurality of frames, training a generative model having a contention relation with the classification model based on the GAN, estimating lost packet information based on the trained generative model with respect to the voice signal encoded by a codec, and restoring a lost packet based on the estimated packet information.

A method for using a waveform segment in place of a missing portion of an audio waveform generated in response to a packet stream encoding portions of the audio waveform, the method comprising: phase matching a trailing portion of the waveform segment with a trailing portion of the audio waveform that follows the missing portion; and adding the phase matched waveform segment to the audio waveform.

A method for performing packet loss concealment of lost packets in Voice over IP (Internet Protocol) speech transmission. Statistical n-gram models are initially created with use of a training speech corpus, and then, packets lost during transmission are advantageously replaced based on these models. In particular, the existence of statistical patterns in successive voice over IP (VoIP) packets is advantageously exploited by first using conventional vector quantization (VQ) techniques to quantize the parameter data for each packet with use of a corresponding VQ index, and then determining statistical correlations between consecutive sequences of such VQ indices representative of the corresponding sequences of n packets. The statistic n-gram predictive models so created are then used to predict parameter data for use in representing lost data packets.