107 results about "Packet loss concealment" patented technology

The present invention discloses a method for obtaining an attenuation factor. The method is adapted to process the synthesized signal in packet loss concealment, and includes: obtaining a change trend of a pitch of a signal; obtaining an attenuation factor, according to the change trend of the pitch of the signal. The present invention also discloses an apparatus for obtaining an attenuation factor. A self-adaptive attenuation factor is adjusted dynamically by using the latest change trend of a history signal by using the present invention. The smooth transition from the history data to the data last received is realized so that the attenuation speed is kept consistent between the compensated signal and the original signal as much as possible for adapting to the characteristic of various human voices.

A technique for performing frame erasure concealment (FEC) in a speech decoder. One or more non-erased frames of a speech signal are decoded in a block-independent manner. When an erased frame is detected, a short-term predictive filter and a long-term predictive filter are derived based on previously-decoded portions of the speech signal. A periodic waveform component is generated using the short-term predictive filter and the long-term predictive filter. A random waveform component is generated using the short-term predictive filter. A replacement frame is generated for the erased frame. The replacement frame may be generated based on the periodic waveform component, the random waveform component, or a mixture of both.

An “adaptive audio playback controller” operates by decoding and reading received packets of an audio signal into a signal buffer. Samples of the decoded audio signal are then played out of the signal buffer according to the needs of a player device. Jitter control and packet loss concealment are accomplished by continuously analyzing buffer content in real-time, and determining whether to provide unmodified playback from the buffer contents, whether to compress buffer content, stretch buffer content, or whether to provide for packet loss concealment for overly delayed or lost packets as a function of buffer content. Further, the adaptive audio playback controller also determines where to stretch or compress particular frames or signal segments in the signal buffer, and how much to stretch or compress such segments in order to optimize perceived playback quality.

The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.

The invention discloses a method for packet loss concealment and a system therefor. The method comprises the following steps: according to the periodicity of a pitch and a phase of a signal, a section of time domain signal is taken in a previous frame and a later frame of a lost signal respectively for the conversion from time domain to frequency domain to obtain a previous frame frequency domain coefficient and a later frame frequency domain coefficient; amplitude values of the previous frame frequency domain coefficient and the later frame frequency domain coefficient are interpolated to obtain amplitude values of frequency domain coefficients of a plurality of reconstruction signals; phases which are most similar to those of the reconstruction signals are taken in the previous frame or the later frame respectively to serve as phase values of the frequency domain coefficients of the reconstruction signals; the conversion from frequency domain to time domain is performed according to the amplitude values of the frequency domain coefficients of the reconstruction signals to obtain time domain signals of the reconstruction signals, and time domain signals of the reconstruction signals are superposed to restore the lost signal. The adoption of the method can improve the accuracy of the period of the reconstruction signals and the matching precision of the phase.

The present invention is directed to voice communication devices in which an audio stream is divided into a sequence of individual packets, each of which is routed via pathways that can vary depending on the availability of network resources. All embodiments of the invention rely on an acoustic prioritization agent that assigns a priority value to the packets. The priority value is based on factors such as whether the packet contains voice activity and the degree of acoustic similarity between this packet and adjacent packets in the sequence. A confidence level, associated with the priority value, may also be assigned. In one embodiment, network congestion is reduced by deliberately failing to transmit packets that are judged to be acoustically similar to adjacent packets; the expectation is that, under these circumstances, traditional packet loss concealment algorithms in the receiving device will construct an acceptably accurate replica of the missing packet. In another embodiment, the receiving device can reduce the number of packets stored in its jitter buffer, and therefore the latency of the speech signal, by selectively deleting one or more packets within sustained silences or non-varying speech events. In both embodiments, the ability of the system to drop appropriate packets may be enhanced by taking into account the confidence levels associated with the priority assessments.