HK1097083B - Realizing high quality lpcm audio data as two separate elementary streams - Google Patents

Description

Implementing high quality LPCM audio data in two separate elementary streams

Technical Field

The present invention relates generally to the field of digital sound recording, and more particularly to editing (editing) digital audio content to support two or more audio formats of different quality.

Background

As the value and use of information has increased, individuals and businesses have sought other ways to process and store information. One option that users may select is an information handling system. An information handling system generally processes, compiles, stores, and/or communicates information or data for business, personal, or other purposes thereby allowing users to take advantage of the value of the information. Because of the differences in technology and information processing needs and requirements among different users or applications, information handling systems may also vary as to what information is processed, stored, or communicated and how quickly and efficiently the information may be processed. Variables in an information handling system allow the information handling system to be diverse and configurable for a particular user or for a particular use (e.g., financial transaction processing, airline reservations, corporate data storage, or global communications). Additionally, an information handling system may include a variety of hardware and software components configured to process, store, and communicate information and may include one or more computer systems, data storage systems, and networking systems. Information handling systems are continually increasing in the performance of hardware components and software applications to generate and manage information.

One application that has developed the fastest for the use of information handling systems is audiovisual systems, particularly those involving High Definition Television (HDTV). With the increasing popularity of HDTV, consumer demand for prerecorded, high definition video and audio content has increased rapidly. The need to match audio quality to high definition video has resulted in new digital audio editing formats using Linear Pulse Code Modulation (LPCM) such as with a sampling rate of 192KHz and 24 bit sampling precision (samplesize), referenced below with respect to LPCM 192/24.

Editing audio content in LPCM192/24 high definition format produces large data files, especially when multiple channels are encoded. The need to accommodate these large files has resulted in larger capacity formats such as "high definition" DVD (HD-DVD) and "Blu-Ray", both of which use blue lasers to read and write digital content. The original DVD capacity is limited to the single layer format 4.7GB and the dual layer format 8.4 GB. HD-DVD has a capacity of 15GB per layer, while blu-ray is capable of delivering 25GB per layer. In the dual-layer version, the two formats can provide 30GB and 50GB of capacity, respectively. These higher capacity media appear to provide a solution to accommodate the necessary large file sizes that are closely tied to high definition video and audio content.

However, reference digital media players, as well as new players that use older digital-to-analog converters (DACs) and smaller capacity digital processors, cannot interpret LPCM192/24 content in its original mode. In order to make the content distributed in the LPCM192/24 format downward compatible, mandatory support of a secondary audio track in a standard digital audio format is required. The new disc format specifies LPCM96/24, dolby digital (AC-3) or DTS (digital cinema system) 5.1 as a Mandatory (directory) secondary audio track which under current implementations contains a secondary audio track in addition to the LPCM192/24 bitstream.

With current implementations of the new optical disc format, forcing the secondary audio track to be recognized by the reference media player and extracted for processing, LPCM96/24, with a 96KHz sampling rate and 24 bit sampling precision, is the preferred format for forcing the secondary audio stream as it passes next to the highest audio quality of LPCM192/24 and is readable by all reference players. Dolby digital and DTS (digital cinema system) 5.1 provide poor quality because they encode at 48KHz/16 bits and 48KHz/20 bits, respectively, and are standards for lossy compression.

Current implementations including LPCM96/24 and LPCM192/24 bitstreams produce disproportionately large file sizes. When combined with high definition video content, these large files can produce a combined file size that exceeds the capacity of the optical disc. For example, the LPCM192/24 audio format for six channels (left, center, right, left rear, right rear, and low frequency) requires 27 Mbps. High compressed high definition video requires 6 Mbps. A mandatory secondary audio channel supporting 96KHz and 24 bits under the current implementation requires an additional 14.4Mbps, resulting in a total of 47.4 Mbps. The 25GB blu-ray DVD can only support such a combined bit rate of 70 minutes.

Currently, for content editing, the most common solution is to embed a lower quality forced audio stream (e.g., dolby digital or DTS) that reduces the post-edited audio file size to help fit all the required content within the capacity limitations of the optical disc. When the method supports the need for LPCM192/24 to support mandatory, secondary audio formats, it limits the audio quality available to users that can decode higher quality mandatory audio formats (e.g., LPCM96/24), but not the media player of LPCM 192/24.

As can be seen from the above, the system requires that a higher quality audio format (e.g., LPCM96/24) be provided to a user of a reference media player while still providing LPCM192/24 files without exceeding the capacity of the media.

Disclosure of Invention

The present invention overcomes the deficiencies of the prior art by a method and apparatus for splitting an LPCM192/24 bitstream into two elementary streams (elementary streams). In one embodiment of the invention, the primary bitstream is in LPCM96/24(96KHz sampling rate and 24 bit sampling precision) format that can be synthesized by media players into a mandatory audio format. The secondary bitstream includes additional bits needed to support the LPCM192/24 format. Media players that are only capable of providing LPCM96/24 format can operate by providing the primary bitstream in its original format. A player capable of synthesizing an LPCM192/24 format combines the primary and secondary bitstreams to produce a combined LPCM192/24 bitstream for synthesis. The combined size of the resulting primary and secondary bitstream files is smaller than the file size generated by the current implementation of the LPCM192/24 enabled secondary audio stream LPCM 96/24. Using the method and apparatus of the present invention, high definition formats can be supported with reduced file sizes, and reference media players will be able to synthesize the highest quality audio formats they can support.

Drawings

The present invention may be better understood, and its numerous objects, features, and advantages made apparent to those skilled in the art by referencing the accompanying drawings. The same reference numbers will be used throughout the drawings to refer to the same or like elements.

FIG. 1 is a diagrammatic view of an information handling system for carrying out the method and apparatus of the present invention;

FIG. 2 is a diagrammatic view of a method and apparatus for editing audio content into a dual-stream LPCM192/24 format;

fig. 3 is a more detailed diagram showing how the present invention splits the original LPCM192/24 into two resultant bitstreams.

Fig. 4 illustrates another embodiment of the present invention that produces slightly lower fidelity.

Detailed Description

FIG. 1 is a diagrammatic view of an information handling system 100 for performing the methods and apparatus of the present invention. The information handling system includes a processor 102, input/output (I/O) devices 104 such as a display, keyboard, mouse, and associated controllers, a hard disk drive 106 and other storage devices 108 such as floppy disks and other storage devices, and various other subsystems 110, all interconnected by one or more buses 102. In one embodiment of the present invention, subsystem 110 includes an optical disc system 114, and optical disc system 114 includes an optical disc 116, and optical disc 116 contains a plurality of data streams that are processed to produce a high quality audio signal, as discussed in more detail below. As discussed in detail below, one of the bit streams is a mandatory, downward compatible format that can be processed by a digital-to-analog (DAC) converter 118, while the other bit stream is an alternative higher quality format that can be processed by a DAC 120. The video data bit stream from the optical disc 116 is processed by the video DAC 122.

For purposes of this disclosure, an information handling system may include any instrumentality or aggregate of instrumentalities operable to compute, classify, process, transmit, receive, retrieve, originate, store, display, manifest, detect, record, reproduce, handle, or utilize any form of information, intelligence, or data for business, scientific, control, or other purposes. For example, an information handling system may be a personal computer, a network storage device, or any other suitable device and may vary in size, shape, performance, functionality, and price. The information handling system may include Random Access Memory (RAM), one or more processing resources such as a Central Processing Unit (CPU) or hardware or software control logic, Read Only Memory (ROM), and/or other types of nonvolatile memory. Additional components of the information handling system may include one or more hard disk drives, one or more network ports for communicating with external devices as well as various input and output (I/O) devices, such as a keyboard, a mouse, and a video display. The information handling system may also include one or more buses operable to transmit communications between the various hardware components.

Fig. 2 is a diagrammatic view of the data structures implemented in the present method and apparatus for editing audio content into a dual-stream LPCM192/24 format. In various embodiments of the invention, the data format illustrated in FIG. 2 is capable of supporting a high quality (e.g., LPCM96/24) mandatory audio format, but consumes less storage space than current implementations of MPCM192/24 having a mandatory secondary audio format of the same quality.

During the digital audio editing process, two bitstreams 200 and 210 are produced from the same audio content. In one embodiment of the invention, the bitstream 200 is a mandatory audio format that needs to be supported and includes consecutive (and ongoing) frames 202, 204 of audio content sampled at 96KHz and read in 24-bit words. The bit stream 210 includes successive (and ongoing) frames 212, 214, 216, 218 sampled at 192 KHz. However, alternate (and proceeding) frames 212, 214 can be written as 0-bit length words and alternate (and proceeding) frames 214, 218 can be written as 24-bit length words.

In this embodiment, a media player capable of synthesizing only LPCM96/24 format identifies a LPCM96/24 bitstream 220, the bitstream 220 including consecutive (and ongoing) frames 222, 224 decoded by the mandatory format DAC118 shown in FIG. 1. In this embodiment, a media player capable of synthesizing an LPCM192/24 format includes merging bitstreams 200 and 210 into a single bitstream 230 in real time, the bitstream 230 including successive (and ongoing) 192KHz-24 bit frames 232, 234, 236, 238, the bit frames 232, 234, 236, 238 then being synthesized by the optional high quality DAC120 shown in FIG. 1.

As discussed in more detail below, the present invention can support multiple audio formats to produce a mandatory, primary audio stream with a significantly reduced post-edit file size compared to the implementation of the currently optional LPCM192/24 format. Those skilled in the art will appreciate that the present invention applies equally to reducing the bandwidth required to transport audio files for network transmission.

Fig. 3 is a more detailed diagram showing how the present invention splits the original LPCM192/24 into two resultant bitstreams. To maintain sample-to-sample synchronicity between the primary and secondary audio streams, the audio content 300 must first be edited into the original LPCM192/24 bitstream 310. In one embodiment of the present invention, editing the original LPCM192/24 bitstream 310 uses 96KHz (f) with low pass_c) An analog-to-digital converter (ADC)302 of a cut-off smoothing (anti-alias) filter. The original LPCM192/24 bit stream 310 includes "n" consecutive 192KHz/24 bit frames. The number of frames starting from the primary frame 312 to the next to the last frame is targeted as "n-1" frame 316, with half of the frames designated as "odd" and the frame 316 designated as "odd". Starting with the second frame 314 and proceeding to the last frame 318, labeled as frame number "n", the other half of the LPCM192/24 frames 330 are designated as "even".

In one embodiment of the invention, the intermediate primary 96KHz, 24-bit audio bitstream 320 is derived from the original LPCM192/24 bitstream 310 is extracted to meet the requirements of the mandatory audio format. The intermediate primary 96KHz-24 bit audio bitstream 320 is generated from odd samples 322, 324 up to the last odd sample 326 labeled frame "n-1" of the original LPCM192/24 bitstream 310. The resulting intermediate primary 96KHz-24 bit audio bitstream 320 is fed through to a (f) stream having 48KHz_c) To a low pass frequency filter 340 for smoothing. The filtered 96KHz-24 bit audio bitstream 360 is synthesized from the filtered frames 362, 364 and up to the last filtered frame 366 labeled "n-1'".

The second intermediate bit-stream 330 is composed of the remaining even frames 332, 334 up to 336 marked as the number of frames "n". This second intermediate bitstream 330 is used to generate the final 192/24 bitstream 390 by additional processing steps described below.

The filtered 96KHz-24 bit audio bitstream 360 is composed of (f) bits having a frequency of 48KHz_c) The low pass frequency filter 340 produces an even number of frames containing low frequency information. The second intermediate bit stream 330 has a frequency of (f) of 96KHz_c) The second intermediate bitstream 330 is passed by a high-pass frequency filter 350 which is used in combination with an interpolation process to produce a bitstream 370 consisting of an odd number of frames 372, 374 through 376 (labeled frames "n-li") carrying high frequency audio data.

The interpolated sample bitstream 370, containing odd samples with high frequency audio data, can be spliced with the filtered 96KHz-24 bit audio bitstream 360 to produce a forced bitstream 380 of full frequency including full frequency frames 382, 384, and up to the last filtered frame 386 (labeled "n-1 f"). The full primary bitstream 380 can be composed by a media player capable of decoding LPCM96/24 format. The full-frequency primary bitstream 380 can also be combined with the intermediate secondary bitstream 330 to produce a final full-frequency LPCM192/24 bitstream 390, the bitstream 390 comprising full-frequency odd frames 392 through to a final odd frame 396 (labeled "n-1"), and full-frequency even frames 394 through to a final even frame 398 (labeled "n"). Thus, the final full-band LPCM192/24 bitstream 390 is composited by any media player capable of decoding LPCM192/24 format.

FIG. 4 illustratesAnother embodiment of the present invention is shown that produces a slightly lower frequency range than is typically available from the 192KHz sampling frequency, but retains the advantage of low noise due to the higher sampling frequency. To maintain sample-to-sample synchronicity between the primary and secondary audio streams, the audio content 400 must first be edited into the original LPCM192/24 bitstream 410. In one embodiment of the present invention, editing the original LPCM192/24 bitstream 410 uses smoothed 48KHz (f) with a low pass_c) Analog-to-digital converter (ADC)402 of the cut-off filter. The original LPCM192/24 bitstream 140 includes "n" number of consecutive 192KHz-24 bit frames. Half of the frames are designated as "odd", starting with the primary frame 412 and up to the frame 416 immediately adjacent to the last frame, labeled as frame number "n-1". The other half of the LPCM192/24 frames 430 are designated as "even" starting with the second frame 414 and continuing through the last frame 418 labeled as frame number "n".

In one embodiment of the invention, an intermediate primary 96KHz-24 bit audio bitstream 420 is extracted from the original LPCM192/24 bitstream 410 to meet the requirements for providing the mandatory audio format. The intermediate primary 96KHz-24 bit audio bitstream 420 is generated from the odd samples 422, 424 of the original LPCM192/24 bitstream 410 up to the odd sample 426 of the last marked frame "n-1". The second middle 96KHz-24 bit audio bitstream 430 is made up of the remaining even frames 432, 434 and up to 436, the 436 being labeled as the number of frames "n".

The intermediate primary 96KHz-24 bit audio bitstream 420 combines with the second intermediate primary 96KHz-24 bit audio bitstream 430 to produce a final LPCM192/24 bitstream 490, the bitstream 490 including a finite frequency, odd frames 432 up to a final odd frame 436 labeled "n-1", and a finite frequency, even frames 434 up to a final even frame 438 labeled "n". The final LPCM192/24 bitstream 430 can be synthesized by any media player capable of decoding the LPCM192/24 format, but will not produce audio content with all of the spectral components present in current LPCM192/24 implementations.

Use of the present invention will ensure that the high quality mandatory audio formats with reduced file sizes supported by at least a portion of the LPCM192/24 implementation accommodate the limitations of distributed media capacity. Also, a media player that is unable to read audio content in the LPCM192/24 format will be able to synthesize the same audio content in the LPCM192/24 format instead of the poor quality audio format due to media capacity limitations.

Although the present invention has been described in detail, it should be understood that various changes, substitutions and alterations can be made hereto without departing from the spirit and scope of the invention as defined by the appended claims.

Claims (9)

1. A method of generating an audio signal using a storage medium, comprising:

a) storing a data file on the storage medium, the data file comprising a digital representation of an original audio signal;

b) generating primary and secondary elementary data streams from said data file;

c) generating a primary audio signal of a primary audio quality using the primary elementary data stream; and

d) generating a secondary audio signal of a secondary audio quality using the secondary elementary data stream;

wherein said secondary elementary data stream comprises a plurality of data frames containing data sampled at 192KHz with alternate frames of said secondary elementary data stream being read as 0-bit length words and 24-bit words, respectively, and data in said frames containing 24-bit words is combined with data of said primary data stream to produce an audio signal containing a plurality of 192 KHz/24-bit data frames.

2. The method of claim 1, wherein the steps b), c), and d) are performed on the same information handling system.

3. The method of claim 1, wherein said primary base data stream comprises a 96KHz/24 bit digital representation of said original audio signal.

4. The method of claim 3, wherein the primary and secondary elementary data streams are edited as follows:

generating a 192KHz/24 bit data stream corresponding to said original audio signal, said 192KHz/24 bit data stream comprising a plurality of consecutive odd and even 192KHz/24 bit data frames;

using a 48KHz filter to generate a plurality of consecutive odd and even 192KHz/24 bit data frames; and

the successive odd frames are used to generate 96KHz/24 bits representing the original audio signal.

5. The method of claim 4, wherein the plurality of odd and even frames combine to produce a 192KHz/24 bit audio signal.

6. An apparatus for generating an audio signal using a storage medium having stored thereon a data file comprising a digital representation of an original audio signal, the apparatus comprising:

means for generating primary and secondary elementary data streams from said data file;

means for generating a primary audio signal of a primary audio quality using the primary elementary data stream; and

means for generating a secondary audio signal of a secondary audio quality using the secondary elementary data stream,

wherein said secondary elementary data stream comprises a plurality of data frames containing data sampled at 192KHz with alternate frames of said secondary elementary data stream being read as 0-bit length words and 24-bit words, respectively, and data in said frames containing 24-bit words is combined with data of said primary data stream to produce an audio signal containing a plurality of 192 KHz/24-bit data frames.

7. The apparatus of claim 6, wherein said primary elementary data stream comprises a digital representation of said original audio signal at 96KHz/24 bits.

8. The apparatus of claim 6 or 7, wherein the primary and secondary elementary data streams are edited as follows:

generating a 192KHz/24 bit data stream corresponding to said original audio signal, said 192KHz/24 bit data stream comprising a plurality of consecutive odd and even 192KHz/24 bit data frames;

using a 48KHz filter to generate a plurality of consecutive odd and even 192KHz/24 bit data frames; and

the successive odd frames are used to generate 96KHz/24 bits representing the original audio signal.

9. The apparatus of claim 8, wherein the plurality of odd and even frames combine to produce a 192KHz/24 bit audio signal.