System for providing a digital watermark in an audio signal

Details System for providing a digital watermark in an audio signal System for providing a digital watermark in an audio signal

1999-10-20

2003-05-27

Mei, Xu (Department: 2644)

Data processing: generic control systems or specific application

Specific application, apparatus or process

Digital audio data processing system

C381S056000, C380S236000, C380S238000

Reexamination Certificate

active

06571144

ABSTRACT:

BACKGROUND OF THE INVENTION
This invention relates generally to signal processing systems, and more particularly to a signal processing system for providing a digital watermark in an audio signal.
With the advent of computer networks and digital multimedia, protection of intellectual property has become a prime concern for creators and publishers of digitized copies of copyrightable works, such as musical recordings, movies, and video games. Once method of protecting copyrights in the digital domain is to use digital “watermarks.” Digital watermarks can be used to mark each individual copy of a digitized work with information identifying, inter alia, the title, copyright holder, and even the licensed owner of a particular copy. Watermarks can also serve to allow for secured metering and support of other distribution systems of a given media content. In theory, almost any item of information could be encoded and used as a watermark.
Digital watermarks are created by encoding a data signal, hereinafter referred to as the “watermark signal,” “watermark data,” or simply “watermark”, which is then integrated into a larger content signal, hereinafter referred to as the “audio signal”, to create a composite signal. Ideally, the composite signal should contain minimal or no perceptible artifacts of the watermark.
It is known in the art that every audio signal generates a perceptual concealment function which masks audio distortions existing simultaneously with the signal. Accordingly, any distortion, or noise, introduced into the transmission channel if properly distributed or shaped, will be masked by the audio signal itself. Such masking may be partial or complete, leading either to increased quality compared to a system without noise shaping, or to near-perfect signal quality that is equivalent to a signal without noise. In either case, such “masking ” occurs as a result of the inability of the human perceptual mechanism to distinguish between two signal components, one belonging to the audio signal and the other belonging to the noise, in the same spectral, temporal or spatial locality. An important effect of this limitation is that the perceptibility of the noise by a listener can be zero, even if the signal-to-noise ratio is at a measurable level. Ideally, the noise level at all points in the audio signal space is exactly at the level of just-noticeable distortion, which limit is typically referred to as the “perceptual entropy envelope” or “PEE”.
Hence, the main goal of noise shaping is to minimize the perceptibility of distortions by advantageously shaping it in time or frequency so that as many of its components as possible are masked by the audio signal itself. See Nikil Jayant et al.,
Signal Compression Based on Models of Human Perception,
81 Proc. of the IEEE 1385 (1993).
“Perceptual coding” techniques employing the above-discussed principles are presently used in signal compression and are based on three types of masking: frequency domain, time domain and noise level. The basic principle of frequency domain masking is that when certain strong signals are present in the audio band, other lower level signals, close in frequency to the stronger signals, are masked and not perceived by a listener. Time domain masking is based on the fact that certain types of noise and tones are not perceptible immediately before and after a larger signal transient. Noise masking takes advantage of the fact that a relatively high broadband noise level is not perceptible if it occurs simultaneously with various types of stronger signals.
Perceptual coding forms the basis for precision audio sub-band coding (PASC), as well as other coding techniques used in compressing audio signals for mini-disc (MD) and digital compact cassette (DCC) formats. Specifically, such compression algorithms take advantage of the fact that certain signals in an audio channel will be masked by other stronger signals to remove those masked signals in order to be able to compress the remaining signal into a lower bit-rate channel.
One of the deficiencies of conventional systems for adding a watermark to an audio signal is that the watermark is encoded on a single frequency band or channel, such that opportunities for inserting the watermark such that it is masked by the PEE of the audio signal are limited. In addition, there exists no option to provide redundancy; that is, the entire watermark is included only once in the audio signal, such that if any part of it is damaged, it is difficult, if not impossible, to recover. Finally, there is no way to “force” an opportunity such that a minimum time between transmissions of the watermark data can be enforced or to “create” an opportunity where one almost exists by changing the gain of the audio signal.
Therefore, what is needed is an improved system for providing a digital watermark in an audio signal.
SUMMARY OF THE INVENTION
The foregoing problems are solved and a technical advance is achieved by a computer-implemented system for providing a digital watermark in an audio signal. In a preferred embodiment, a audio file, such as a .WAV file, containing an audio signal to be watermarked is processed by an encoder using an algorithm of the present invention herein referred to as the “PAWS algorithm” to determine and log the location and number of opportunities that exist for inserting a watermark into the audio signal such that it will be masked by the PEE of the audio signal. The user can adjust certain parameters of the PAWS algorithm before the audio file is processed. A/B/X testing between the original and watermarked files is also supported to allow the user to undo or re-encode the watermark, if desired.
In particular, the encoder divides the frequency spectrum into seven “critical bands”, each of which includes two carrier frequencies for representing logic 0 and logic 1, respectively. The basic encoding process is as follows. First, the user sets up the desired parameters for the algorithm, including selecting which critical bands are to be active, specifying, in dB, the desired “headroom” between the PEE of the audio signal and the amplitude of the encoded watermark signal transmitted in each active band, and specifying the maximum time between transmissions of the encoded watermark signal.
If the encoding is not being performed in real-time, the user executes a preconditioning phase. During preconditioning, the encoder runs through the entire .WAV file and logs watermark opportunities according to the PAWS algorithm and the parameters specified by the user. In addition, the encoder detects “near-miss” opportunities in the audio signal; that is, points in the audio signal that would constitute opportunities with a small adjustment to the gain. The encoder adjusts the gain of the audio signal at that point to create an opportunity therefrom. The preconditioned audio signal is written back to a .WAV file.
In a preferred embodiment, the watermark is formatted as a frame of 32 characters. During operation, the original or preconditioned .WAV file is input to the encoder, which monitors each active critical band of the audio signal to detect opportunities for inserting watermark data in accordance with the PEE of the signal within the band, as well as the user-defined parameters. The existence and location of each opportunity is logged and the encoder determines how many bytes of the watermark word (a “subframe”) may be transmitted during that opportunity, according to the data rate of that band, by measuring the width of an opportunity and dividing by the data rate, which yields the size of the data transmission. The encoder encodes the watermark using Gaussian Minimal Shift Key (“GMSK”) modulation and incorporates the encoded subframes of the watermark data block into the audio signal at the opportunity.
In one aspect, at each opportunity, a timer is reset to a maximum time between opportunities, which is either a default value or a value selected by a user. If the timer times out before the next opportunity is detected, the encoder “forces” an opportunity by cross-

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