BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to the field of multi-channel audio transmission and methods of selecting and manipulation of a plurality of language options for a multi-channel audio transmission.
2. Description of the Related Art
Technological advancement in the audio industry has expanded beyond stereo systems with a left and right channel. These stereo systems have now been replaced by multi-channel surround sound systems. A typical surround sound system will often include a center channel, at least one right channel, at least one left channel, one right surround sound channel, and one left surround sound channel. The surround sound channels are typically placed behind the user to provide a 360 degree sound experience. Surround sound systems can also include a low frequency effects (LFE) channel to generate low frequency sound effects.
Surround sound configurations can have a varying number of channels. For example, a 5.1 surround sound system will include a center channel, a left channel, a right channel, a left surround sound channel, a right surround sound channel, and a LFE channel. In contrast, a 7.1 system includes all the channels found in the 5.1 system and an additional left and right channel. The extra two channels allow the user to have a more rounded listening experience.
In addition to the audio industry, technological advancement has also allowed the world to become a much smaller place. It is not uncommon for a family in the United States to be watching a Japanese reality show or for a family in Denmark to be watching a French soap opera. This has created an increased need to for broadcasters to provide multiple language transmissions for the same programming. Sporting events such as the Olympics and the World Cup are viewed in a hundred different languages all across the world. Viewers often will only be able to receive one language and often it is the native language of the region and not the preferred language of the local viewer.
For broadcast stations to adapt programming to the local language, the process requires large digital consoles, digital to analog convertors, analog to digital convertors, analog mixers, and the expertise of a mix engineer. Performing these functions can be highly costly in terms of time, equipment space, and sound quality. It is common in the industry of broadcast transmission to provide a secondary audio programming (SAP) that allows the user to select a second predetermined audio language. One drawback to SAP programming is it is often limited to a monaural audio signal. So a user desiring the second language will sacrifice the ability to experience the multi-channel experience provided by the native language programming. Even in the native language, the audio signal received is not always at ideal sound levels. Many times, broadcast stations need the option to adjust the sound levels of the signal without the need to change the language.
There is a need for a simpler method for broadcast stations to change the language options of the programming and to adjust the levels of the sound mix without the added expense of time, equipment space, and sound quality.
SUMMARY OF THE INVENTION
The present invention provides a process and system for managing multi-channel audio signals and a plurality of language signals, and decoding the signals into serial sound data to create a program serial data and a plurality of language serial data. The program serial data and the plurality of language serial data are aligned, and the program serial data is separated. The plurality of language serial data are separated to create a plurality of language channels. At least one language channel is mixed with at least one serial data to generate a language channel mix. The levels of each program serial data and language channel mix are adjusted to generate a final output mix. The final output mix is encoded to adhere to the AES-3id standard to create an output signal, and the output signal is then transmitted.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 is a high level block diagram of a surround sound mode of a digital audio routing system according to an embodiment of the present invention;
FIG. 2 is a high level block diagram of another stereo sound mode of a digital audio routing system according to an embodiment of the present invention;
FIG. 3 is a graphical illustration of the audio mixer in the digital audio routing system of FIGS. 1 and 2;
FIG. 4 is a graphical illustration of the oscillator tone generator in the digital audio routing system of the present invention;
FIG. 5 is a block diagram of the components in a digital audio routing system configured in accordance with the present invention; and
FIG. 6 is a high level block diagram of another mono sound mode of a digital audio routing system according to an embodiment of the present invention.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT
Reference will now be made to the drawings wherein like reference designators refer to like components or processes throughout. FIG. 1 is a high level block diagram of a surround sound mode of a digital audio routing system adapted to provide a broadcaster with the ability to transmit different dialog options to a user.
In the surround sound embodiment illustrated in FIG. 1, the system comprises the steps of receiving an incoming surround sound signal 103 from a remote broadcast 101. A transceiver 501 (FIG. 5) can be used as a receiver and transmitter for all audio signals. The signal 103 will follow the AES-3id standard which uses the same cabling, patching, and infrastructure as analogue or digital video, and is thus common in the broadcast industry. The AES-3id standard uses 75-ohm BNC electrical pair connections to enter the receiver. In the illustrated embodiment, the transceiver 501 will accept seven AES pair connections, three AES pairs for the audio inputs and four AES pairs for the language inputs. Once the signal 103 is received, transceiver 501 (FIG. 5) will decode 105 the AES-3id signals into Integrated Interchip Sound (IIS) serial data interface. IIS is an electrical serial bus interface standard used for connecting integrated circuits in an electronic device. The decoded signals 105 will contain separate 106 program serial data and language serial data as shown in FIG. 1.
The program serial data and language serial data will be aligned 107 to a master clock using a sample rate converter 503 (FIG. 5). This step synchronizes all the audio signals. Synchronization is necessary because not all signals use the same sampling rates. For example, American television (48 kHz), European television (44.1 kHz), and movies (48 kHz or 96 kHz) all use different sampling rates. Just replaying the existing data at the new rate will not normally work, since it introduces large changes in pitch for audio, plus it cannot be done in real time. In the broadcast industry, separate devices in a broadcast studio function at different sample rates. Additionally, the sample rates may be the same, but there may be timing differences between devices. Examples of the devices include but are not limited to CD players, tape machines, computers, and asynchronous satellites. The sample rate converter 503 (FIG. 5) can change the sampling rate while changing the information carried by the signal as little as possible.
Once aligned, the program data and language data can be injected with an oscillator tone (FIG. 4) using an oscillator 405, equalizer 406, and an oscillator multiplexer 407. The oscillator tone 408 is used for testing purposes. The oscillator tone (FIG. 4) is injected to allow a broadcast engineer to confirm the routing path of the data and verify that a signal is being received. The program data and language data will then be separated 109/111 (FIG. 1). In surround sound mode shown in FIG. 1, the program data is separated into a center speaker channel, left speaker channel, right speaker channel, left surround speaker channel, and right surround speaker channel 122. In the stereo mode of FIG. 2, the program data will separate 109 to a left speaker channel and a right speaker channel 122. The language data will be separated 111 into a maximum of eight different language channels 112. A plurality of audio multiplexers 509 and language multiplexers 511 (FIG. 5) will select the inputs to be sent to a plurality of mixers 513. There is one mixer 513 for each separate language channel 112 (FIG. 1).
Each mixer 513 (FIG. 3) will have three signal inputs, the desired broadcast language 301, the original native language 303, and the auxiliary signal 305, as well as individual level controls 300. The mixer 513 will combine signals to create a language channel mix 307. In surround sound mode FIG. 1, the center speaker channel is used in the mixer 513. In stereo mode FIG. 2, both the left speaker channel and the right speaker channel 122 will process through the mixer 114. In certain embodiments, the auxiliary signal 305 (FIG. 3) may contain dialog placed on top of the original language dialog. This may include narration from varied viewpoints such as color commentary, play by play perspective, or additional dialog separate from the original signal. For example, in one embodiment, the auxiliary signal 305 can allow the broadcaster to say “up next on the local news” during the credits of a television show. After each mixer 513, the signal again goes through an oscillator 505 and multiplexer 507 (FIG. 5) for testing/signal verification purposes. The language channel mix 307 (FIG. 3) is added with the program channels 122 (FIG. 1) to create a final output mix 120 which is sent to be encoded 117.
The levels of the language channel mix 307 (FIG. 3) are adjusted 115 via a touch screen interface 515 (FIG. 5), rotary interface, or remote ethernet interface. The ethernet interface allows parameter adjustment over a computer network. The interface can also adjust the levels of the program data and language data 121 when each is separated 109/111. The language channel mix 307 (FIG. 3) is added with the program channels 122 (FIG. 1) to create a final output mix 120 which is sent to be encoded 117.
In the FIG. 2 embodiment of the stereo mode of operation, the program separation step 109 into left and right channels 122 takes place simultaneously with the separation 111 of the language signals. The language channels 112 go through a mono to stereo split 116. The mono to stereo split 116 will divide each language channel 112 into a left language channel and a right language channel 118. Once the levels of the left channel and right channel 118 are adjusted 115, the left language channel and the right language channel 118 are sent to the mixer 513 (FIG. 5) for the step of combining the signals. Accordingly, the left channel 122 is mixed with the left language channel 118 and the right channel 122 is mixed with the right language channel 118 to create a left channel mix and a right channel mix 114 of the program and language signals. The left channel mix and right channel mix 114 are added together to create the final output mix 120 which will be sent to be encoded 117.
In the FIG. 6 embodiment of the mono sound mode of operation, the program serial data 609 is mixed with at least one language channel 112 to form an output mix 120 which will be sent to be encoded 117.
Once the final output mix is encoded back to the AES-3id standard 117 (FIGS. 1, 2), the mix is sent back to the transceiver 501 (FIG. 5) to be transmitted 119 to the appropriate location.
Patent Grant
9350474
