An audio digital CODEC can be connected to a plurality of digital transmission facilities. The CODEC has a plurality of programmable compression schemes which are upgradeable and downloadable. One of the programmable compression schemes is provided with various parameters that when changed affect the quality of the resultant audio. These psycho-acoustic parameters include the standard ISO parameters and additional parameters and can be monitored and controlled by a user.
RELATED APPLICATION
This application is a continuation of U.S. patent application Ser. No. 08/630,790, now U.S. Pat. No. 6,041,295, filed Apr. 10, 1996, which is a continuation of U.S. patent application Ser. No. 08/420,721, filed Apr. 10, 1995 now abandoned and a continuation of U.S. patent application Ser. No. 08/419,200, filed Apr. 10, 1995, now abandoned.
A system for recognizing the existence of and adjusting the psycho-acoustic parameters present in an audio digital CODEC. A audio digital CODEC is provided with various parameters that when changed affect the quality of the resultant audio. These psycho-acoustic parameters include the standard ISO parameters and additional parameters to aid in effecting a pure resulting audio quality. The psycho-acoustic parameters located in the audio digital CODEC can be monitored and controlled by the user. The parameters can be monitored by a speaker associated with the CODEC or headphones. The user can control the adjustment of the psycho-acoustic parameters through the use of knobs present on the front panel of the CODEC or graphic or digital representations. Adjustment of the parameters will provide real time change of the resulting audio sound that the user can monitor through the speaker or the headphones. Dynamic Psycho-acoustic parameter Adjustment (DPPA) permits the user to dynamically change the values of different parameters. The ability to change the parameters can be embodied in front panel knobs or in the action of computer software as instructed by the user.
In one embodiment, the invention is directed to methods and system for converting an analog signal to digital samples for transmission over a communication network, and for converting digital samples received over a communication network to an analog signal. According to one feature, the system of the invention generates encoding and decoding master clocks from local oscillators, thus enabling the system of the invention to operate in environments where reliable timing signal are not available from the communication network. According to another feature, the system of the invention adjusts the frequencies of the encoding and decoding master clocks based on a connect rate to the communication network. In a further feature, the system of the invention employs encoding and decoding buffers for buffering the digital samples between a modem or a digital network access device, and signal converters to maintain a defined time relationship between digital samples being transferred between the modem or the digital network access device and the signal converters.
A method of encoding a multi-channel audio signal including at least a first signal component (LF), a second signal component (LR) and a third signal component (RF). The method comprises the steps of encoding the first and second signal components by a first parametric encoder (202) resulting in a first encoded signal (L) and a first set of encoding parameters (P2); encoding the first encoded signal and a further signal (R) by a second parametric encoder (201), resulting in a second encoded signal (T) and a second set of encoding parameters (P1), where the further signal is derived from at least the third signal component; and representing the multi-channel audio signal at least by a resulting encoded signal (T) derived from at least the second encoded signal, by the first set of encoding parameters and by the second set of encoding parameters.
Methods and apparatus for broadcasting high quality audio "studio direct" with the same digital information employed in the studio by the video producer with AC-3 digital audio signals for broadcast to integrated receiver decoders (IRD). The methods and apparatus permit proper handling of AC-3 data by switching signals to encoders in response to detection of the encoded signals representing compression of the data. Control over individual data bits such as copyright bits is maintained by determining the bit status, comparing it to a preferred status, changing the status if it does not comply with the preferred status, and reevaluating cyclical redundancy check value in each data packet to avoid disruption in the data transmission. In addition, the system includes an uplink device which automatically checks, logs and reports errors in Dolby Digital AC-3 signals by a monitor which employs a processor, a digital audio card and an SMPTE timecode reader. As an option, an ethernet interface may be provided to permit AC-3 transmission to expedite storage and transmission of the audio data by media such as compact disks. The monitor employs a state machine that finds AC-3 packets, locks into the packets and detects discontinuities or loss of signal. The monitor then computes and checks the cyclical redundancy check value of the AC-3 packet found. In addition, the system enables the device to play AC-3 signals such as Dolby Digital out in sync with video signals, regardless of the storage media for the files. A sound card having an input for receiving house reference AES clock pulses enables the AES clock of the playback signal to be locked to the frequency of a production house master as a time code reader or an editor's contact closure match video and audio signals playback.