Audio Engineering - Digital Audio

Niquist theory applies to Digital Audio. This theory states that in order to accurately capture and reproduce sound, the sampling rate must be at least twice the highest frequency present in the audio signal. This ensures that all the necessary information is captured and there is no loss of quality during the recording and playback process. MIDI and Analogue Audio do not directly relate to Niquist theory, and Kinetic Energy is unrelated to audio recording.

The correct answer is "The recorded wave form is sampled 48,000 times per second." In a digital audio system, the sample rate determines the number of samples taken per second to represent the analog audio waveform. In this case, with a sample rate of 48kHz, the waveform is sampled 48,000 times per second. This allows for accurate representation of the audio signal and is a common sample rate used in professional audio recording and playback.

The highest frequency that can be recorded without aliasing occurring is equal to half of the sample rate. In this case, since the sample rate is 48kHz, the highest frequency that can be recorded without aliasing occurring is 24kHz.

FLAC (Free Lossless Audio Codec) is a popular audio format that employs lossless compression, meaning it reduces the file size without sacrificing any audio data. This makes it ideal for archiving and preserving high-quality audio recordings. MP3 and AAC, on the other hand, use lossy compression, which discards some audio information to achieve smaller file sizes. WAV is an uncompressed format, resulting in larger file sizes.

An anti-aliasing filter is used to prevent aliasing, which is the distortion or artifacts that can occur when a signal is sampled or digitally processed. A low pass filter allows frequencies below a certain cutoff point to pass through while attenuating frequencies above that point. By using a low pass filter as an anti-aliasing filter, it can effectively remove or reduce high-frequency components that may cause aliasing when converting analog signals to digital or during digital signal processing.

The given correct answer is an accurate possible chain of events for Analogue to Digital Conversion. It starts with the Audible Signal, which is captured by a Microphone and converted into an Analogue Signal. The Analogue Signal then goes through a Low Pass Filter to remove any unwanted high-frequency components. After that, it is converted into a Digital Signal using an A/D Converter, which represents the continuous Analogue Signal as discrete digital values. Therefore, the chain of events provided in the answer accurately represents the process of converting an Audible Signal into a Digital Signal through Analogue to Digital Conversion.

The bit depth refers to the number of bits used to represent the amplitude of a sound wave. In this case, the bit depth is 16, which means that there are 16 bits available to represent the amplitude. Each bit can have two possible values (0 or 1), so when you raise 2 to the power of 16, you get 65,536. This means that there are 65,536 different levels of amplitude that can be recorded with a bit depth of 16.

Bit depth refers to the number of bits used to represent each sample in a digital audio file. It determines the resolution and dynamic range of the audio. A higher bit depth allows for a greater number of possible values for each sample, resulting in a more accurate representation of the original sound. Therefore, the correct answer is "States the number of different values each sample can be."

If the bit depth is 8, it means that each sample in the recording can have 8 bits of information. Since each bit can have two possible values (0 or 1), the total number of different amplitude levels that can be recorded is 2^8, which equals 256.

The range of dynamic levels (dynamic range) is primarily determined by the bit depth, not the sample rate. Increasing the sample rate does not directly affect the dynamic range; it mainly affects the frequency resolution and fidelity of the recorded audio.

Aliasing occurs when a frequency higher than the Nyquist frequency (half the sample rate) is sampled, causing it to be incorrectly represented as a lower ("ghost") frequency in the digital signal. This is a sampling error, not related to bit depth, and it is prevented by using a low-pass (anti-aliasing) filter, not a high-pass filter.

In order to accurately replay a wave form at the correct frequency, it must be sampled at least twice the frequency of the wave. This is known as the Nyquist-Shannon sampling theorem. Therefore, the minimum amount of times a wave form must be sampled is twice the frequency, which is represented by option 4.

To avoid aliasing, the highest frequency must be half the sample rate according to the Nyquist Theory. This theory states that in order to accurately record a waveform without distortion, the highest frequency present in the signal must be sampled at least twice per cycle. Therefore, the sample rate must be at least twice the highest frequency to prevent aliasing, where higher frequencies are incorrectly represented as lower frequencies. By setting the highest frequency to be half the sample rate, the Nyquist Theory ensures that the signal is properly sampled and avoids aliasing.

An A/D Converter running at 48kHz, 16bit does not measure the frequency of the waveform 48,000 times per second. Instead, it measures the amplitude of the waveform 48,000 times per second and stores it as a 16bit word. The converter turns the analogue signal into a string of 1's and 0's, generating 768,000 bits per second.

The settings for a Red-Book Audio CD include a sample rate of 44,100hz, a bit depth of 16, a maximum length of 74 minutes, and a maximum number of tracks of 99. These settings are specified by the Red Book standard for audio CDs, which ensures compatibility with CD players. The sample rate and bit depth determine the quality of the audio, with 44,100hz being the standard sample rate for audio CDs and 16 being the standard bit depth. The maximum length and maximum number of tracks are limitations set by the Red Book standard to ensure that the CD can be played on any CD player.