Description of DC Live/Forensics Software Program

DC Live/Forensics is a real-time, effective and easy-to-use software program designed for analyzing degraded audio recordings and includes an extensive suite of filtering modules for enhancing speech intelligibility and for reducing distracting and fatiguing to listen to background noise and interfering sounds found on most forensic tape recordings.

The program has the tools necessary to analyze an audio signal to locate problems and to enable the user to process a noisy, hard-to-understand tape recording and produce an enhanced copy with noticeable improvement.

Every spoken word is important when a recording is being used to establish or substantiate what was said in a conversation. An unintelligible group of words could contain a statement that is important and could even be key to the actual meaning of the conversation. Even if a recording is intelligible, DC Live/Forensics can be used to reduce distracting and tiring background noise and to restore the correct frequency response using Equalization (EQ) making the enhanced copy easier to listen to.

DC Live/Forensics uses the computer's CPU to efficiently process digital audio in real-time. When the software is installed and running on the user's computer system, it provides over 20 precise and effective filter modules and advanced signal analysis features to enhance the intelligibility of hard-to-understand recordings.

By studying the User's Manual and understanding the logic and concepts behind DC Live/ Forensics and letting go of old procedures and hardware, DC Live/Forensics will provide the user with a new, faster, and better way of processing degraded audio recordings to enhance speech intelligibility.

To assist the user in understanding and using DC Live/Forensics, the program package includes a printed User's Manual that is over 350 pages.

Real-Time Processing

DC Live/Forensics has been optimized to "run" in real-time. When the operator/examiner changes a filter setting, the results of this change are heard immediately. Real-time processing enables the operator/examiner to select the optimum filter settings by adjusting the controls while listening.

Each of the filter modules has a "default" set-up and several set-ups that are optimized to handle common problems. Using the default or one of the custom settings as a starting point, the operator listens to the processed audio output and decides if the processing is satisfactory. If additional enhancement is needed, the operator/examiner adjusts each filter control in the direction that improves the intelligibility of the spoken words and reduces interfering noise until the intelligibility of the spoken words begins to get worst and/or the interfering noise(s) gets louder. The operator/examiner then moves the control back to the best position.

In addition, each filter module has a "by-pass" button that enables the operator/examiner to make an A/B comparison of the original and processed audio to accurately assess the effectiveness of the enhancement.

Time Domain and Frequency Domain Analysis Suite

Click on each item for screen shot

The Time Domain window displays time on the "X" axis and the amplitude of the audio signal on the "Y" axis. This display can be used by the operator/examiner to visually locate and identify - clicks, pops, transients, fades in volume level, signal drop-outs, far-party/near-party level differences, and other signal amplitude changes.

An A/B comparison of the Time Domain of the original and processed signals can be displayed titled horizontally on the computer monitor in time sync. This enables the operator/examiner to quickly determine if the filtering modules are effectively correcting/reducing amplitude (level) problems on the audio being processed by observing the "before and after" Time Domain waveforms. (Click Here to View an A/B Comparison Time Domain Display)

Frequency Domain

The Frequency Domain window displays frequency on the "X" axis and the amplitude of the frequency components on the "Y" axis. This display is called an FFT and can be used by the operator/examiner to visually locate and identify - tones, hum, low frequency "rumble" and wind noise, high frequency hiss, nulls and resonances in the audio signal caused by the room acoustics when the recording was made, interference caused by the microphone and recording channel electronics, and the frequency response (shape) of the frequency spectrum of the audio signal. The FFT can be used to analyze the input signal and then switched to the output (processed signal) to verify filter performance.

Spectrogram

In addition, the program can produce a spectrogram display of the audio. A spectrogram displays time on the "X" axis, frequency on the "Y" axis and amplitude (loudness) as intensity. This display is sometimes called a "voice print." It is useful in identifying continuous tones or bands of frequencies and can be used to analyze the voice of a speaker.

X-Y Vector Display

Used to measure the phase between two audio signals. Usually the left and right channels of a stereo signal. One audio signal is fed into the "X" input and the second audio signal is fed into the "Y" input. The amplitude of the "X" audio signal is the x value of the vector and the amplitude of the "Y" audio signal is the y value of the vector at each moment in time.

An X-Y Vector Display that is at 45 degrees and has a positive slope (rising), indicates that the two Waveforms are time aligned and in phase.

Filtering Modules
Click on filter to see screen shot
Impulse Filter: EZ / Expert	Automatically locates and corrects for clicks, pops, and transients. Has controls that can be adjusted in real-time to optimize detection and correction of these types of interference.
Low Pass Filter:	Used to reduce the frequency components of the recording that are "above" the limited voice frequency spectrum (200 to 3,200 Hertz). When set to 3,200 Hertz, reduces above voice spectrum hiss and interference without affecting the limited voice spectrum.
High Pass Filter:	Used to reduce the frequency components of the recording that are "below" the limited voice frequency spectrum (200 to 3,200 Hertz). When set to 200 Hertz, reduces low frequency rumble, wind noise and the lower frequency hum components without affecting the limited voice spectrum.
Band Pass Filter:	A combination low pass and high pass filter. When set to 200 and 3,200 Hertz respectively, reduces "noise" components that are "above" and "below" the limited voice frequency band (200-3,200 Hertz).
Notch Filter:	Used to reduce continuous tones that are inside or outside of the voice frequency spectrum.
Harmonic Reject Hum Filter:	Used to reduce a set of hum components that are integer multiples.
Graphic Equalizer 10 Band:	Provides 10 or 20 bands (sliders) that are adjusted to compensate for nulls and resonances in the frequency spectrum and to reduce tones and out of voice spectrum frequencies.
Paragraphic Equalizer 10 Frequency Point:	Used as graphic equalizer but the frequency points are selectable and adjustable by the operator.
Continuous Noise Filter:	Used to reduce stationary noise components across the entire spectrum of the audio signal. The operator/examiner locates a short segment of the audio signal that does not contain speech and whose noise is representative of the noise heard throughout the audio recording. This "noise only" segment is used to make a "noise print." The continuous noise filter removes this "noise print" from the entire recording (or segment). This filter is effective in removing all noises and interferences that are contained in the "noise print."
Adaptive Filter:	Also called a one-channel or two-channel adaptive filter. The adaptive filter automatically builds a model of the stationary and slowly changing interference and noise components present in the audio being processed and then removes this interference/noise model from the audio signal. This is a very powerful enhancement tool.
Spectral Filter:	Provides up to 32,000 bands of equalization. The operator/examiner can precisely (to less than one Hertz resolution) reduce, accent, shape, cut, and boost any frequency in the audio spectrum. A custom display is provided (along with default set-ups) to assist the operator in obtaining optimum performance from this filter.
Brick Wall Filter(s):	The set of Brick Wall Filters are optimized for Forensic work where high roll-off transition bands are required. Four types of filters are provided - low pass, high pass, band pass and band stop. These filters are used when the operator/examiner needs to precisely remove/reduce noise components that are very close to the "desired" signal.
Multi Filter:	The multi filter is one of the powerful features provided by the DC Live/ Forensics software program. The multi filter provides the operator/examiner with a signal path between an input source and an output destination. The input source can be the "live" input from an A/D converter (Sound Card) or a stored input from a Wave File. The output destination can be the "live" output of a D/A converter (full duplex sound card required) or sent to the computer's hard drive for storage. Any filter module can be selected with the mouse and placed in the signal path. The signal path can accommodate up to 256 filters. Any combination of filter modules can be inserted in the signal path. The same filter module may be repeated in the signal path. This allows the operator/examiner to increase the effect of a single filter module by inserting two or more of the same filter modules in the signal path. The repeat filter modules may have the same or different parameter settings. Each filter module has a set of user adjustable controls for precisely and effectively adjusting its parameters to enhance degraded speech recordings in real-time. The multi filter enables the operator/examiner to design, test, evaluate, set-up and apply a complex set of filters that have been selected and optimized to solve the multiple problems found on forensic recordings. The multi filter does not convert the digital data stream that is being processed back to Wave file resolution (16 or 24 bits) between each filter module in the signal path but maintains a signal path that has the precision of the arithmetic of the software program. This prevents the accumulation of rounding errors. In other words, it is better to use the multi filter anytime more than one filter module will be used.

DC Live/Forensics has additional filters and capability that are explained in the User's Manual.

In summary, DC Live/Forensics provides an extensive collection of the analysis tools needed to accurately locate, identify, and characterize the noise components of forensic recordings and the necessary filter modules to effectively reduce these noise components.

JBR Technology is the U. S. Distributor for DC Live/Forensics. JBR Technology has extensive experience in forensic audio and recording equipment and is available to provide assistance in using and understanding this effective software program. If you have any questions or need additional information, please contact jim@jbrtech.com or 703-644-5500.

Pricing:

The current law enforcement price for the DC Live/Forensics program is $1,399. Delivery is within seven (7) days after receipt of order. Shipping and handling is $8.00. Free Technical Support is provided for 6 months from 8:00 a.m. to 4:30 p.m. Monday through Friday at support@tracertek.com.