Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion
Reexamination Certificate
1999-03-22
2001-10-30
Dorvil, Richemond (Department: 2641)
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
C704S212000
Reexamination Certificate
active
06311161
ABSTRACT:
TECHNICAL FIELD
The present invention relates generally to digital data decoding systems, and more particularly, to a system and method for mixing multiple digital audio streams into a single analog output.
BACKGROUND OF THE INVENTION
Full motion video and audio displays based on digital signals have become widely available. While these displays have many advantages, they also often require a massive amount of raw digital data. Because the storage and transmission of digital video and audio signals is central to many applications, and because an uncompressed representation of a video and audio signal requires a large amount of storage, the use of digital compression techniques is vital to this advancing art.
Several international standards for the compression of digital video and audio signals have emerged over the past decade, with more currently under development. These standards apply to algorithms for the transmission and storage of compressed digital data in a variety of applications, including: video-telephony and teleconferencing; high quality digital television transmission on coaxial and fiberoptic networks; as well as broadcast terrestrially and other direct broadcast satellites; and in interactive multimedia products on CD-ROM, digital audio tape, and Winchester disk drives.
Several of these standards involve algorithms based on a common core of compression techniques, e.g., the CCITT (Consultative Committee on International Telegraphy and Telephony) Recommendation H.120, the CCITT Recommendation H.261, and the ISO/IEC MPEG-1 and MPEG-2 Standards. The MPEG algorithms have been developed by the Moving Picture Experts Group (MPEG), part of a joint technical committee of the International Standards Organization (ISO) and the International Electro-technical Commission (IEC). The MPEG Committee has been developing standards for the multiplex, compressed representation of video and associated audio signals.
Briefly summarized, the MPEG-2 standard describes an encoding method that results in substantial bandwidth reduction by a subjective lossy compression followed by a lossless compression. The encoded, compressed digital data is subsequently decompressed and decoded in an MPEG-2 compliant decoder.
Video and audio decoding in accordance with the MPEG-2 standard is described in greater detail in commonly assigned U.S. Pat. No. 5,576,765, entitled “Video Decoder”, as well as commonly assigned pending application Ser. No. 09/046,283, entitled “Method To Play Audio And Video Clips Through MPEG Decoders”. This patent and application are hereby incorporated herein by reference in their entirety.
Audio/video decoders are typically embodied as general or special purpose processors and memory. Decoders that are used with television sets are often referred to in the industry as set-top box (STB) systems.
Stand-alone audio/video decoders currently used in STB systems generally utilize a dedicated interface to a specific transport chip and/or network interface module (NIM). As the required utility of these STB systems expands, it is becoming desirable to adopt input from various sources in addition to the base network (e.g., cable/satellite) connection.
For example, in addition to a main audio stream, such as a soundtrack for a movie, it may be desirable for the transport stream carrying the program to also contain a secondary audio stream or channel that provides, e.g., additional narrative for disabled viewers who are unable to hear subtle sounds in the original soundtrack. Alternatively, it may be desirable to allow a user to run other functions in the background and request audio queues when certain events have occurred, such as receiving mail, etc. Further, it may be desirable to allow an internet browser or other interactive application to run in the foreground, with primary audio and video programming (e.g., a movie) running in a window on the display. In such cases, it is desirable that the secondary or supplementary audio sound be mixed with the original audio stream so that both can be heard at a single time through a common set of speakers, as opposed to mixing the two streams where only one or the other is heard.
As an additional consideration, in order to accomplish decoding of multiple audio streams, the base audio decoder function would conventionally be replicated for as many digital audio streams as output is desired. However, this could be costly and unnecessary in many potential audio mixing applications such as those noted above.
Therefore, in order to establish commercial advantage, there is a need for an integrated device capable of decoding and merging multiple independent digital audio streams using a single base audio decoder. The present invention is directed to meeting this need.
DISCLOSURE OF THE INVENTION
Briefly described, the present invention comprises in one aspect a system for mixing multiple audio streams, including a primary stream of audio data and a secondary stream of audio data. The system has an audio decoder for decoding the primary stream of audio data and system memory for holding the secondary stream of audio data as audio pulse code modulated (PCM) data. A controller retrieves the secondary stream of audio data for forwarding to an audio playback macro for formatting. The system also includes a mixer for mixing the output of the audio decoder and the formatted secondary stream of audio data from the playback macro as a merged audio signal.
In another aspect, a method for mixing multiple audio streams is presented which includes: decoding a primary stream of audio data and producing therefrom a decoded primary stream of audio data; retrieving a secondary stream of audio data from system memory, the secondary stream comprising audio pulse code modulated (PCM) data; formatting the secondary stream of audio data for mixing with the decoded primary stream of audio data; and mixing the decoded primary stream of audio data and the formatted secondary stream of audio data for output as a merged audio signal.
To restate, provided herein is a device/method for taking one or more secondary audio channels from either a streaming application or from memory, and playing the secondary channels out in PCM format concurrently with output from a base audio decoder for analog mixing thereof into a single audio stream. This single audio stream is achieved without the need for multiple base audio decoders or other external components to create the second digital audio output stream to be mixed with the primary stream.
The solution presented herein advantageously employs a generic programming interface. The second audio stream can be either a static file in memory or result from a streaming application, e.g., received as network input. In addition, the secondary audio stream can be received directly as a PCM file, or it can be the result of processing (decoding, etc.) by the system processor. Still further, since two or more audio streams are converted to analog format and then mixed in the analog domain, there is no restriction on the sample frequency associated with the original digital input, and no complex sample rate conversion is required.
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IBM Technical Disclosure Bulletin, “Resource Management System for Multimedia Devices”, vol. 36, No. 09B, (Sep. 1993), pp. 525-529.
Anderson Richard Eugene
Foster Eric M.
Franklin Dennis Edward
Dorvil Richemond
Heslin & Rothenberg, P.C.
International Business Machines - Corporation
Radigan, Esq. Kevin P.
Steinberg, Esq. William H.
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