摘要:
A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The transmitter predicts the number of encoded frames, Fpred, in the buffer having a limited level and transmits the value, Fpred, to the receiver with the frame. If the transmitter determines that the decoder buffer level is high, the frames being generated by the encoder are small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming low, the frames being generated by the encoder are big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver clock using feedback depending on the compared level of the decoder to the actual level to Fpred.
摘要:
One aspect of the invention relates to statistically multiplexing first services and second services in a group. A measure of required bandwidth for the first services is obtained, where the first services comprise pre-encoded services. An available encoding bandwidth for the second services is determined from a group bandwidth for the first and second services using the measure of required bandwidth. An encoding bit rate is allocated to each of the second services based on the available encoding bandwidth. Each of the second services is encoded in accordance with the encoding bit rate thereof. One or more services of the first services and the second services are transcoded and a multiplex is formed. Since the available encoding bandwidth for the second services is determined using the measure of required bandwidth for the first services, transcoding of the second services is minimized, and video quality is maximized.
摘要:
Here is described a method, a near end telecommunications terminal and a computer executable software code which allow to perform a telecommunications with a far end telecommunications terminal providing the option of applying a narrowband encoding technique without loosing the benefit of a wideband encoding technique. This is achieved after setting up a telecommunications between that near end telecommunications terminal and the far end telecommunications terminal by applying by the codec from the near end telecommunications terminal a sampling when encoding corresponding to a wideband encoding technique while assembling the resulting timeslots into frames to be transmitted to the far end telecommunications terminal via RTP. Furthermore, the sampled timeslots are assembled when applying the wideband encoding technique according to the parity of the sequential integer number from the sampled timeslots to generate two kind of frames, one with all the timeslots identified by even sequential numbers, the other one with the remaining timeslots identified by odd sequential numbers.
摘要:
A method and apparatus are disclosed for controlling a buffer in a digital audio broadcasting (DAB) communication system. The decoder buffer level limits are specified in terms of a maximum number of encoded frames (or duration). The transmitter can predict the number of encoded frames, Fpred, in the decoder buffer and transmit the value, Fpred, to the receiver with the audio data. If the transmitter determines that the decoder buffer level is becoming too high, the frames being generated by the encoder are too small and additional bits are allocated to each frame for each of the N programs. Likewise, if the transmitter determines that the decoder buffer level is becoming too low, the frames being generated by the encoder are too big and fewer bits are allocated to each frame for each of the N programs. The transmitted predicted buffer level, Fpred, can also be employed to (i) determine when the decoder should commence decoding frames; and (ii) synchronize the transmitter and the receiver. The receiver fills the decoder buffer until Fpred frames are received before commencing decoding frames when the decoder first starts up or possibly when a new audio program is selected. The transmitter and receiver clocks may be synchronized by adjusting the clock at the receiver by using a feedback loop that compares the actual level of the decoder buffer to the predicted value, Fpred, received from the transmitter (a higher number of encoded frames in the buffer indicates that the clock of the receiver is too slow and should be increased, and a lower number of encoded frames in the buffer indicates that the clock of the receiver is too fast and needs to be slowed down).
摘要:
Techniques to adjust clock approximations are described, which may be used to synchronize content output at a client. In an implementation, timestamps derived from a universal time source are allocated to respective program clock reference (PCR) timestamps in content received by a network operator during an interval of time to form ordered pairs of timestamps. An approximation is computed of a plurality of the ordered pairs of timestamps for the interval and the approximation is adjusted using an ordered pair of timestamps taken from a previous approximation.
摘要:
A signal transmitter and a signal receiver comprises a time base compression part (101) for compressing an audio signal on a time base to output the compressed signal as a time base compressed audio signal and a multiplexing part (102) for multiplexing a video signal, a control signal, and the time base compressed audio signal to output the multiplexed signal to the external as a video/audio control multiplexed signal. The thus constituted signal transmitter and the signal receiver realize a signal transmission system for multiplexing and transmitting the video signal, audio signal, and control signal, for consequently transmitting high-quality diverse digital audio signals without conversion into an analog signal, and for transmission at high speed with minimal error.
摘要:
A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.
摘要:
A signal processing system which discriminates between voice signals and data signals modulated by a voiceband carrier. The signal processing system includes a voice exchange, a data exchange and a call discriminator. The voice exchange is capable of exchanging voice signals between a switched circuit network and a packet based network. The signal processing system also includes a data exchange capable of exchanging data signals modulated by a voiceband carrier on the switched circuit network with unmodulated data signal packets on the packet based network. The data exchange is performed by demodulating data signals from the switched circuit network for transmission on the packet based network, and modulating data signal packets from the packet based network for transmission on the switched circuit network. The call discriminator is used to selectively enable the voice exchange and data exchange.
摘要:
A system and method for provide a stable gain from an adaptive gain control device in a signal path. An echo canceller is also located in the signal path, and is used to provide performance information regarding losses in the signal. This performance information is fed to the automatic gain control device via a connection. The automatic gain control device thereafter uses the performance information to determine a maximum gain that might be provided based upon losses cause by echo conditions. The gain however is limited in order to provide for a stable system. The performance information includes a loss rate that includes a combination of the echo return loss and the echo return loss enhancement.
摘要:
Disclosed are a statistical multiplex system, a statistical multiplex controller and a method of statistical multiplex, which can assign bit rates to program data and auxiliary data efficiently, and also improve image quality. A statistical multiplex system is provided with: a plurality of image encoders for encoding a plurality of program data; an information encoder for encoding the auxiliary data; a multiplexing apparatus for multiplexing output thereof, and a statistical multiplex controller for controlling each of the image encoders and the information encoder. The statistical multiplex controller is made to set the bit rate to be assigned to the information encoder first, and to assign the remaining bit rates to each of the image encoders.