Abstract: A demodulator (10) converts television signals to video baseband signals and audio baseband signals including stereo signals representing a right channel signal value, a left channel signal value and a pilot signal having an amplitude. A DSP (60) recursively finds a coefficient value for scaling that keeps the pilot signal amplitude within a range of values, there by improving baseband signal quality, such as stereo separation of stereo signals within the baseband signals.

Abstract: A system and method are disclosed for performing digital multi-channel decoding of a BTSC composite audio signal. Each subsequent stage of the digital multi-channel decoding process is performed at the lowest sampling rate that yields acceptable performance for that stage. Analog-to-digital conversion of the composite audio signal is performed first to generate a composite digital audio signal. After analog-to-digital conversion, all signal processing may be performed in the digital domain. The composite digital audio signal is digitally filtered to frequency compensate for variations caused by previous stages of processing, including IF demodulation. Digital channel demodulation and filtering are performed to isolate single channels of the composite digital audio signal such as SAP, L−R, and L+R channels.

Abstract: A demodulator (10) converts television signals to video baseband signals and audio baseband signals including stereo signals representing a right channel signal value and a left channel signal value. A DSP (60) recursively finds a preferred coefficient value for a scaling that reduces stereo separation due to amplitude variation of the right and left channel signal values. The preferred coefficient value is thereafter used for scaling the right and left channel signal values.

Abstract: A method and apparatus are disclosed for performing sampling rate conversion of an audio signal from a first sampling rate to any of a plurality of higher sampling rates using a single set of low-pass filter coefficients. Sampling rate conversion is accomplished by effectively up-sampling, low-pass filtering, and down-sampling the audio signal to generate interpolated output samples of a second digital audio signal at any of a plurality of sampling rates. The sampling rate conversion process includes storing a fixed set of filter coefficients as a plurality of phased subsets of filter coefficients, applying samples of the audio signal to the phased subsets in a rotational manner to generate filtered samples of the audio signal, and selecting and linear interpolating between certain filtered samples to generate samples of the second digital audio signal.

Abstract: Certain embodiments of the present invention provides a system and method for SAP FM demodulation. The system includes a bandpass filter for isolating the SAP signal, a Hilbert filter to produce a copy of the SAP signal phase shifted by 90 degrees, an FM demodulator for demodulating the SAP signal using the phase shifted SAP signal and a delayed SAP signal, and a lowpass filter to eliminate noise from the FM demodulated SAP signal. The system may also include an automatic gain control for normalizing amplitude of FM demodulator input signals. The digital FM demodulator uses a simplified approximation using non-unity delay for simplified demodulation of frequency modulated signals.