摘要:
A transmission system with a transmitter and a receiver. The transmitter has a speech encoder with analysis means, has calculation means, and has control means. The receiver has a speech decoder. Through a transmission medium, the transmitter transmits frames of data to the receiver. The analysis means determine analysis coefficients from a speech signal. From a bitrate setting, the calculation means calculate a fraction of the frames of data to carry more information about the analysis coefficients than a remaining number of the frames of data. The control means control the transmitter to transmit the fraction of the frames of data and to transmit the remaining number of the frames of data. The receiver receives the frames of data. The receiver derives a reconstructed speech signal from the received frames of data.
摘要:
A narrow-band audio signal (9) contains information, present as recognisable distortions, for processing the signal into a wide-band signal. In a method for processing a wide-band audio signal (1) into a narrow-band signal (9), a first spectral portion (4) is maintained unchanged in the narrow-band signal and information (7) usable for restoring remaining spectral portions (5) is embedded (8), preferably perceptually inaudible, by distorting said first spectral portion in a recognisable way, and preferably as a watermark. An encoder for coding a wide-band audio signal (1) into a narrow-band signal (9) and a decoder for decoding a narrow-band audio signal are disclosed, as well as a system for transmitting a wide-band audio signal through a narrow-band transmission channel, a system for storing a wide-band audio signal on a storage medium and retrieving the wide-band signal from storage, and a storage medium carrying a narrow-band audio signal.
摘要:
The invention concerns audio coding methods and particularly relates to an efficient means by which selected frequency bands of information from an original audio signal which are audible but which are perceptually less relevant need not be encoded, but may be replaced by a noise filling parameter. Those signal bands having content which is perceptually more relevant are, in contrast fully encoded. Encoding bits may be saved in this manner, without leaving voids in the frequency spectrum of the received signal. In this way, this method avoids the annoying bandwidth switching artefacts that can occur when full bandwidth audio is encoded with a bit budget which is too low to represent the signal within each frequency band. Thus, this method allows an increase in the encoded audio bandwidth without introducing annoying bandwidth switching artefacts. The noise filling parameter is a measure of the RMS signal value within the band in question and is used at the reception end by a decoding algorithm to indicate an amount of noise to inject in the frequency band in question.
摘要:
In a communication system, a multimedia signal is encoded in an encoder (1) and subsequently transmitted over a packet switched network (4) to a terminal (6). The terminal (6) comprises a receiver (8) whose output is connected to a receive buffer (10). The output of the receive buffer (10) is applied to the presentation means (14) which comprises a decoder (16) and a presentation device (18). In order to deal with delay variations in the packet switched network (4), it is proposed to change the presentation speed of the multimedia signal dependent on the transmission delay of the multimedia signal. This is done by a controller (12) that determines the number of packets in the buffer (10) and adapts the decoding rate and the playback rate of the multimedia signal accordingly.
摘要:
In a speech communication network in which a transmitter (1) transmits speech signals via a network (4) to a receiver (8), it can happen that more traffic is offered to the network (4) than it can handle. In order to reduce the network load, at least one node (24) comprises bitrate reduction means to delete some of the prediction parameters representing the speech signal. It can also be the case that a receiver comprises a speech decoder having not the computational power available for decoding the encoded speech signal. In said case, the speech decoder is arranged for using only a part of the prediction parameters available. This results in a lower complexity of the synthesis filter (60).
摘要:
In a speech communication network in which a transmitter (1) transmits speech signals via a network (4) to a receiver (8), it can happen that more traffic is offered to the network (4) than it can handle. In order to reduce the network load, at least one node (24) comprises bitrate reduction means to delete some of the prediction parameters representing the speech signal. It can also be the case that a receiver comprises a speech decoder having not the computational power available for decoding the encoded speech signal. In said case, the speech decoder is arranged for using only a part of the prediction parameters available. This results in a lower complexity of the synthesis filter (60).
摘要:
The invention concerns audio coding methods and particularly relates to an efficient means by which selected frequency bands of information from an original audio signal which are audible but which are perceptually less relevant need not be encoded, but may be replaced by a noise filling parameter. Those signal bands having content which is perceptually more relevant are, in contrast fully encoded. Encoding bits may be saved in this manner, without leaving voids in the frequency spectrum of the received signal. In this way, this method avoids the annoying bandwidth switching artefacts that can occur when full bandwidth audio is encoded with a bit budget which is too low to represent the signal within each frequency band. Thus, this method allows an increase in the encoded audio bandwidth without introducing annoying bandwidth switching artefacts. The noise filling parameter is a measure of the RMS signal value within the band in question and is used at the reception end by a decoding algorithm to indicate an amount of noise to inject in the frequency band in question.
摘要:
In an audio transmission system, an input signal is split up into two spectral portions in a transmitter. These spectral portions are coded by their own respective coder. The low-frequency signal portion is coded by a regular narrow-band coder and the high frequency portion is coded using a coder that outputs LPC codes and signal amplitude codes. In the receiver, the low frequency signal portion is reconstructed by a narrow-band decoder and the high frequency portion is reconstructed by applying a high pass filter to a white noise signal and applying an LPC filter that is controlled by the LPC codes to this filtered white noise signal and adjusting the signal amplitude with an amplifier that is controlled using the amplitude codes of the transmitter. The reconstructed low frequency signal and the reconstructed high frequency signal are then combined to yield a reconstructed output signal containing both frequency ranges.
摘要:
A communication system (1) comprises a transmitter (2), a receiver (3), and an up/down link communication channel (4, 6) arranged for data communication from the transmitter (2) through the up link communication channel (4) to the receiver (3). The communication system (1) is further arranged to feedback data from the receiver (3) through the down link communication channel (6) to the transmitter (2). The receiver (3) comprises a bad frame indicator (5) for providing a bad frame indication (BFI) upon receipt of a corrupted frame, which is present in synchronized data communicated over the up link communication channel (4); and the transmitter (2) comprises resynchronization means (7) coupled to the down link communication channel (6) for receiving BFI related data and in response thereto recommencing data communication over the up link communication channel (4), in accordance with a resynchronization procedure, which starts from a predetermined state. A fast acting feedback resynchronization procedure for a GSM speech system is presented which prevents substantial error propagation from occurring at the receiver end.