Signal Theories
Being in the VoIP business, it's imperative to be familiar with the various theorems in dealing with digital voice signals.
Sampling Theory
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Samples are successive snapshots of a signal. In the case of audio, the signal is a sound wave. A microphone converts the acoustic signal into a corresponding analog electric signal, and an analog-to-digital converter transforms the analog signal into a sampled digital form. The accuracy of the digital approximation of the analog signal depends on its resolution in time (the sampling rate) and its quantisation, or resolution in amplitude (the number of bits used to represent each sample). for example, the audio recorded for storage on compact disc is sampled 44 100 times per second and represented with 16bits per sample.
Telephone speech is sampled at 8kHz. 16kHz is generally regarded as sufficient for speech recognition and synthesis. The audio standard is a sample rate of 44.1kHz (Compact Disc) or 48 kHz (Digital Audio Tape) to represent frequencies up to 20 kHz.
We use the term sample to refer to a single output value from an A/D converter, i.e., a small integer number (usually 8 or 16 bits). Audio data is characterized by the following parameters, which correspond to settings of the A/D converter when the data was recorded. Naturally, the same setting must be used to play the data.
- sampling rate (in samples per second) 8000 or 44100
- number of bits per sample 8 or 16
- number of channels ( 1 for mono, 2 for stereo)
The correct term is samples/sec or samples per second not Hz or kHz.
Sampling rates are always measured per channel, so for stereo data
recorded at 8000 samples/sec, there are actually 16000 samples in a second.
If you want to create an Oscilloscope type signal from your input device then you need to apply this theorem.
Nyquist Theorem
=============
Check out the rest here
Sampling Theory
============
Samples are successive snapshots of a signal. In the case of audio, the signal is a sound wave. A microphone converts the acoustic signal into a corresponding analog electric signal, and an analog-to-digital converter transforms the analog signal into a sampled digital form. The accuracy of the digital approximation of the analog signal depends on its resolution in time (the sampling rate) and its quantisation, or resolution in amplitude (the number of bits used to represent each sample). for example, the audio recorded for storage on compact disc is sampled 44 100 times per second and represented with 16bits per sample.
Telephone speech is sampled at 8kHz. 16kHz is generally regarded as sufficient for speech recognition and synthesis. The audio standard is a sample rate of 44.1kHz (Compact Disc) or 48 kHz (Digital Audio Tape) to represent frequencies up to 20 kHz.
We use the term sample to refer to a single output value from an A/D converter, i.e., a small integer number (usually 8 or 16 bits). Audio data is characterized by the following parameters, which correspond to settings of the A/D converter when the data was recorded. Naturally, the same setting must be used to play the data.
- sampling rate (in samples per second) 8000 or 44100
- number of bits per sample 8 or 16
- number of channels ( 1 for mono, 2 for stereo)
The correct term is samples/sec or samples per second not Hz or kHz.
Sampling rates are always measured per channel, so for stereo data
recorded at 8000 samples/sec, there are actually 16000 samples in a second.
If you want to create an Oscilloscope type signal from your input device then you need to apply this theorem.
Nyquist Theorem
=============
Check out the rest here
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