Abstract:
A method for performing audible fast-forward or reverse of audio content represented in a compressed format, such as, but not limited to, MPEG-1 Layer 3 (MP3) or MPEG-2 Advance Audio Coding (AAC) employs a fast-forward controller which performs fast-forward or reverse by repeatedly skipping forward or reverse in the compressed audio data stream, retrieving a block of data, and then splicing these data blocks back together. A decoder is then used to decode each of these blocks, to detect when a block switch has occurred (a splice in the data stream), and to quickly resynchronize at each transition. Hierarchical or multiplexed data streams may be decoded using a cascade of decoders each employing this technique. The decoder uses a robust sync search for performing resynchronization and error recovery.

Description:
This application claims priority under 35 USC §119(e)(1) of Provisional Application No. 60/170,449, filed Dec. 13, 1999. 
    
    
     TECHNICAL FIELD OF INVENTION 
     The present invention relates to method and apparatus for use of digital communications signals, and more particularly, to method and apparatus for audible fast-forward or reverse of compressed audio content. 
     BACKGROUND 
     Providing an audible fast-forward/reverse support for compressed audio formats is typically a challenge due to resynchronization issues associated with such formats. FIGS. 6 and 7 show the formats of a typical compressed audio data stream. The data stream is divided into units called frames. Each frame represents a segment of audio data which can be decoded. At the start of each frame is a header, which contains general information about the data stream, i.e., sampling rate, bit rate, profile, etc. The first word of the header is a syncword, which is a string of bits that identifies the “start” of a frame. 
     For current generation personal, digital audio players, the fast-forward and reverse functions are silent. That is, the user only hears silence during the fast-forward (or reverse) operation. A simple fast-forward technique involves jumping forward in the data stream by an amount associated with the desired fast-forward rate, and then re-synchronizing based on the frame headers. To do this resynchronization, a search may be employed which searches the data stream for the string of bits which matches a syncword. When this syncword is found, the decoder can begin parsing the frame. In addition, a cyclic redundancy check (CRC) can be used to detect errors in the data stream. 
     However, even this simple technique has many practical problems associated with it. Many compressed audio file formats, such as MPEG-1 Layer 3 (MP3) or MPEG-2 AAC, utilize large amounts of variable length coding. Certain allowable sequences of these variable length codes can actually emulate the syncword, i.e., the syncword is not unique. It is a common occurrence to find a match to a “false” syncword when searching in such a data stream. 
     In addition, the CRC check is typically not required, and is therefore not always transmitted. When it is transmitted, significant parsing and/or computation may be required to determine the validity of the frame. In some audio transports, a 1-bit field is used to indicate whether the CRC is used. Parsing a faulty header which has this bit set to 0 (caused by faulty syncword detection) may, in fact, cause the CRC check to be disabled. 
     Because of the use of variable-length codes, parsing errors can occur (and be decoded/played) undetected, since even random noise can sometimes be a valid sequence of codewords. This results in the output of the decoder being badly distorted. 
     Furthermore, in some audio data streams, such as those associated with MP3, the data required to decode the frame actually occurs before the syncword and header. A field in the header tells the decoder where to look for data. This pointer will always point backwards, and in some cases will even point to a location before the previous frame hear. When a break or discontinuity in the data stream occurs, resynchronization is difficult because, even though the header has been found, the data may not be complete. 
     Synchronization of hierarchical or multiplexed data streams poses an additional problem. In such data streams, an outer bitstream which may be encrypted carries pieces of an inner data stream as payload. This multiplexed data stream may be decoded by an outer decoder that extracts the payload data and supplies it to an inner decoder. Following a splice in the data stream, the outer decoder/decryptor must first gain synchronization, followed by the inner decoder. This compounds the difficulty of problems such as resynchronization time and error robustness. 
     SUMMARY 
     The present invention provides a method and apparatus for audible fast-forward or reverse of compressed audio content. 
     The present invention provides a method for performing audible fast-forward/reverse of audio content represented in a compressed format, such as, but not limited to, MPEG-1 Layer 3 (MP3) or MPEG-2 Advance Audio Coding (AAC). A fast-forward controller is employed which performs fast-forward or reverse by repeatedly skipping forward or reverse in a stored compressed audio data stream, retrieves a chunk of data, and then stores these data chunks in an end-to-end fashion, such that they are spliced back together as a single data stream. A decoder is then used to decode each of these chunks, to detect when a chunk switch has occurred (a splice in the data stream), and to quickly resynchronize at each transition. Hierarchical or multiplexed data streams may be decrypted and decoded using a cascade of decoders each employing this technique. The decoder uses a novel and robust syncword search for performing resynchronization and error recovery. 
    
    
     BRIEF DESCRIPTION OF DRAWINGS 
     For a more complete understanding of the present invention, and the advantages thereof, reference is now made to the following detailed description taken in conjunction with the accompanying drawings, in which: 
     FIG. 1 depicts a block diagram of a hardware platform for playing digital audio files for an individual end user, e.g. a portable audio player, capable of implementing the methods of the present invention; 
     FIG. 2 depicts a high level block diagram of a software hierarchy suitable for use on the hardware platform of the present invention depicted in FIG. 1; 
     FIG. 3 depicts a simplified flow diagram of the fast-forward (or reverse) method of the present; 
     FIG. 4 depicts a simplified flow diagram for one synchronizing method of the present invention; 
     FIG. 5 depicts a simplified flow diagram for a synchronizing method of the present invention as applied to MP3 audio format; 
     FIG. 6 depicts a block diagram of a frame structure for an AAC audio format; and 
     FIG. 7 depicts a block diagram of a frame structure for an MP3 audio format; 
    
    
     DETAILED DESCRIPTION 
     The present invention provides a method and apparatus for method and apparatus for audible fast-forward or reverse of compressed audio content. 
     Referring now to FIG. 1, there may be seen a hardware platform or system  100  for playing digital audio files for an individual end user, e.g. a portable audio player. The digital audio files may be in various compressed and/or encoded formats. This platform  100  is contained within a suitable holder to allow for easy transport, operation, and replacement of stored audio files (audio content). 
     As may be seen from FIG. 1, the platform includes a digital signal processor (DSP)  110  and a microcontroller  120  that are interconnected. Typically, the DSP  110  executes a stored program  112 ,  114 , or  116  for decoding and decompressing stored audio data files. The microcontroller  120  provides appropriate displays  122  and controls keypad  124  to allow the user to operate the platform. Both the microcontroller  120  and DSP  110  may include on-chip memory  132 ,  130  for storing the programs for operating the microcontroller  120  and DSP  110  to provide desired functionality for the portable audio player. The platform  100  also includes a flash memory  140 , which is preferably removable, for storing the digital audio data. 
     The platform includes batteries  150 , which are replaceable and/or rechargeable, that supply power for the other devices on the platform. There is a power supply block  160  associated with and connected to the batteries  150 . The power supply block  160  includes a DC to DC converter  162  for converting the voltage of the batteries to those voltages required to operate the devices on the platform. A voltage regulator  164  may be provided to regulate selected voltages to a desired level of regulation. In addition, a voltage supervisor  166  may be provided as part of the power supply to oversee and control the operation of the DC to DC converter and voltage regulator. For ease of depiction purposes, FIG. 1 does not depict all the interconnections between the power supply and the devices in the platform  100 . 
     The DSP  110  provides an audio bit stream to a stereo digital-to-analog (DAC) converter  170  which converts the digital signals to an analog equivalent. A power amplifier  180  is interconnected with the DAC  170  for amplifying the signal and providing the signal to an output device such as speakers, a set of earphones, or some other device for converting the electrical signal to an audible signal. 
     Preferably, the DSP  110  in addition to decoding also performs equalization, volume, tone, and balance control functions  185  responsive to control signals from the microcontroller as a result of user interactions with control keys in the keypad  124 . Alternatively, the power amplifier may be responsive to control signals from the microcontroller  120  (or the DSP  120 ) for volume, tone, and balance control. 
     The platform preferably includes a crystal  186  for controlling the clock frequency for the DSP  110  and may include a separate crystal for controlling the clock frequency for the microcontroller  120 . Alternatively, the DSP  110  may supply a clock signal to the microcontroller  120 , or vice versa. 
     The flash memory  140  contains the audio files, e.g. audio content, available for listening to by the end user. The audio files are typically stored as separate files for each song, which may be in different audio formats, such as, for example, but not limited to MP3 and AAC. The flash memory  140  may also contain stored program files for decoding each type of audio format, as well as control other operations associated with the methods of the present invention. These files are typically stored with a file extension that identifies the format. Thus, a system controller may recognize the format of the next song desired to be played by the user and recognize that the decoder program currently loaded in local memory  130  associated with the DSP  110  may be of a different format; the system controller may then discard the program for the “old” decoder and reload local memory  130  of DSP  110  with the decoder program for the format of the file desired to be played. For each such data file stored in the flash memory  140 , the audio data is typically stored as a continuous sequence of data in adjacent memory locations. Thus, the sequential data stream in memory may be accessed and retrieved for decoding and/or decryption by addressing the corresponding sequential memory address locations and retrieving data blocks of a size corresponding to the output data word width of the specific memory device employed for the flash memory  140 . 
     An external personal computer (PC)  90  may be appropriately connected with the platform  100 . In this manner, audio files may be loaded directly into memory  140  by the PC  90  or into memory  140  via the microcontroller  120 . Alternatively, the PC  90  may be employed to load the audio files into memory  140  when it is removed from platform  100  and inserted into special hardware interconnected with the PC  90  for downloading audio files into the memory  140 , via an appropriate interconnection for memory  140 . 
     Referring now to FIG. 2, there may be seen a high level block diagram of a software hierarchy  200  suitable for implementing the invention of the present invention on the hardware platform  100  of Figure ore particularly, in FIG. 2 there may be seen a data stream  205  stored in the flash memory  140 . A fast-forward controller  210  is a programmable address generator that moves the sequence of data from the memory to a temporary data buffer  220  when operating in “normal” playback mode. The fast-forward controller  210  also operates in a fast-forward mode in response to a user operation of an appropriate fast-forward (or reverse) control on the audio player. 
     When operating in a fast-forward mode, the fast-forward controller segments the sequential data stream into chunks of data (blocks of data) which are separated in time. The time separation between chunks is determined by the desired fast-forward rate. This rate is preferably adjustable and may be set or selected via software or other control signals. The fast-forward controller  210  is preferably implemented as a program and one that executes on the microcontroller  120 ; although, clearly, this program may also execute on the DSP  110 . In any event, the fast-forward controller moves data from memory to the temporary data buffer  220 . 
     The splice detector and synchronizer block  230  detects the transition between each of these chunks in a fast-forward (or reverse) mode, detects when a chunk switch has occurred (e.g. a splice in the data stream), and then quickly resynchronizes (e.g. finds a syncword and its associated header) after each splice. The splice detector and synchronizer block  230  maintains synchronization and performs resynchronization whenever a break or splice in the data stream occurs. The details for performing such a task are described later herein. Once the data stream is synchronized the data is passed to a decoder  240  which decodes the audio data and provides an output audio data stream  250 . 
     Referring now to FIG. 3, there may be seen a simplified flow diagram  300  for the fast-forward (or reverse) method of the present invention. More particularly, the fast-forward controller  210  jumps forwards (or backwards) in the data stream  310  stored in memory. A chunk of data is then forwarded to a temporary buffer  220 . The fast-forward controller again jumps forwards and then forwards another chunk of data to the buffer. This continues in an ongoing fashion. The splice detector and synchronizer block  230  then searches for a syncword at block  320  in at least a portion of the first chunk of data in the buffer. If a syncword is not located, then additional data of the first chunk from the buffer is examined for a syncword at block  330 . If a syncword is located, then the header is examined to determine if a CRC check is to be performed  340 . If no CRC check is to be performed, the data is passed to the decoder  240  for decoding  350 . If not, the CRC check is performed and evaluated against the provided checksum. If the CRC check is passed, then the data is made available to the decoder  240  for decoding  350  in accordance with the audio format of the data file. If the CRC check fails, then a new search is initiated for a syncword  320  to find another header. 
     Referring now to FIG. 4, there may be seen a simplified flow diagram for the synchronizing method of the present invention. More particularly, it may be seen from FIG. 4 that data from the temporary data buffer is initially searched or analyzed for a syncword. If a syncword is not detected more data is retrieved from the buffer to continue to look for a syncword. If a syncword is located, its associated header is parsed to determine several things. One determination is where the next syncword should be located in the data stream. Another determination is where the data associated with the header starts or is located. This is then used to determine the frame length. Additional determinations are made as to the sampling rate, bit rate, profile, layer, and identification fields. 
     Then a jump is then made to the next header via its corresponding syncword. The next header is then parsed to determine the same information noted above. The information from the two headers are then compared for consistency. In parallel with this consistency comparison a jump is made back to the first header. If the information from the two headers is consistent a check is then made of the need to perform a CRC check. If the information is inconsistent, the data from the data stream is advanced and the search for the first syncword and header begins again. In a similar manner, if the CRC check is performed and the checksum is not correct, then. the search for the first syncword and associated header begins again. 
     If the CRC check is passed, then the data is provided to the decoder. This type of double header search is especially useful for formats like those associated with the AAC format. The AAC frame format is depicted in FIG.  6 . As may be seen from FIG. 6, the frame includes an initial syncword which is immediately followed by the header. The header in turn is followed by a portion of fixed length data. The fixed length data is followed by a packet of variable length data. As noted earlier herein, information about the length of the frame is included in the header. 
     FIG. 7 depicts the frame format for the MP3 audio format. Again there is a syncword followed immediately by header. However, for the MP3 format the data may be partially or entirely “in front” of the header. In FIG. 7, header N is in the “middle” of its associated data packet. Thus, the header N has a pointer to where its data starts (main data begin). However, until header N+1 is decoded it may not be certain where data for packet N ends and data for packet N+1 starts. Thus, there is often a need to evaluate two headers to determine the packet length. The remainder of FIG. 7 depicts the situation where  3  headers must be decoded to determine the packet length for packet N+2. Remember that a frame extends from one syncword to the next syncword, so that the data for a particular packet may span one or more frames. 
     Referring now to FIG. 5, there may be seen a simplified flow diagram for the synchronizing method of the present invention as modified for the MP3 format. The “totalamountMainData” is a variable that serves to capture the length of the data packet after adjustment for (deletion of) headers and syncwords. It is intialized to zero when first starting the syncword search. In a similar manner, the variable “mainDataThisFrame” stores the length of data in the frame under analysis or determination. FIG. 5 adds steps to determine these variables versus headers and frames. 
     For audio data that is encrypted an outer layer is added to the inner layer of data that needs to be decoded as noted earlier herein. These hierarchical or multiplexed data streams may be decrypted and decoded using a cascade of decoders each employing this technique. 
     For splice detection during resynchronization the invention requires that a “candidate” header frame correctly identify the location of the succeeding frame header. For the common case where input data stream data is delivered in sequence to the decoder, the data intervening between these two headers is stored in memory. In the case of a hierarchical data stream with large outer frames, this may impose an unacceptable increase in memory cost beyond what is required to decode the inner data stream. In addition, since the data stream is delivered at a finite data rate to the decoder, the increased memory may cause an increased delay before the outer decoder can make a decision, which may be perceptible in the decoded audio. A further aspect of the invention is a method to minimize the increase in memory and delay in the case of a hierarchical data stream. Details of the method are described later herein. 
     For hierarchical or multiplexed data streams, the syncword search for the outermost encrypted layer may be relaxed to a single syncword and associated header search. This is possible in situations were the encryption is weak in that the data immediately following the encrypted header is encrypted but most of the remaining data in the frame is not encrypted. 
     The present invention is capable of being implemented in software, hardware, or combinations of hardware and software. Although the present invention and its advantages have been described in detail, it should be understood that various changes, substitutions and alterations may be made herein without departing from the spirit and scope of the invention, as defined in the appended claims.