ALBENO P.

ROSANTINA

AIM 2-1

DIGITAL AUDIO THEORY Digital audio is useful in the recording, manipulation, mass-production, and distribution of sound. Modern online music distribution depends on digital recording and data compression. The availability of music as data files, rather than as physical objects, has significantly reduced the costs of distribution.[1] An analog audio system captures sounds, and converts their physical waveforms into electrical representations of those waveforms by use of a transducer, such as a microphone. The sounds are then stored, as on tape, or transmitted. The process is reversed for playback: the audio signal is amplified and then converted back into physical waveforms via a loudspeaker. Analog audio retains its fundamental wave-like characteristics throughout its storage, transformation, duplication, and amplification. A digital audio system starts with an ADC that converts an analog signal to a digital signal.[note 1] The ADC runs at a specified sampling rate and converts at a known bit resolution. CD audio, for example, has a sampling rate of 44.1 kHz (44,100 samples per second), and has 16-bit resolution for each stereo channel. Analog signals that have not already been bandlimited must be passed through an anti-aliasing filter before conversion, to prevent the distortion that is caused by audio signals with frequencies higher than the Nyquist frequency, which is half of the system's sampling rate.A digital audio signal may be stored or transmitted. Digital audio can be stored on a CD, a digital audio player, a hard drive, a USB flash drive, or any other digital data storage device. The digital signal may then be altered through digital signal processing, where it may be filtered or have effects applied. Audio data compression techniques, such as MP3, Advanced Audio Coding, Ogg Vorbis, or FLAC, are commonly employed to reduce the file size. Digital audio can be streamed to other devices.For playback, digital audio must be converted back to an analog signal with a DAC. DACs run at a specific sampling rate and bit resolution, but may use oversampling, upsampling or downsampling to convert signals that have been encoded with a different sampling rate. SOUND In its simplest terms sound is just a vibration that is transmitted through the air to our ears. Most of us have put our hand on a mechanical device that is vibrating and felt the vibration. Normally we also hear the vibration as it affects the air around us. Sound "waves" are successive areas of air compression or rarefaction. The speed of sound is simply the speed at which those areas of compression and rarefaction pass through the atmosphere. Think of what happens when you hit a drum. The air directly under the drumhead is compressed. Next to the area of compression there must be an area with less air- an area of rarefication. This is caused by the drumhead bouncing back up. In fact, the drumhead will vibrate back and forth several times, creating a series of areas that are compressed next to areas that are rarefied. The areas of compressed and rarefied air move out from the drumhead just like ripples on a pond. We call this a sound wave. When it gets to your ears your eardrums move to match the air pressure, and nerves inside your ear pick up the movement and send it to your brain as sound.

HOW IS SOUND DIGITIZED Nearly everything mentioned in the previous tutorials about sound and waveforms applies to digitized sounds or digital waveforms as well. However, because digital waveforms are sampled approximations to continuous functions of time, there are some important differences to be aware of and there is some new terminology to learn. First, digital waveforms are described in terms of their sampling rate in Hertz and sample resolution in bits per sample. The term Hertz, abbreviated Hz, normally means cycles per second when describing waveforms, but it is generalized to samples per second when describing a sampling rate. Each sample in a digital waveform is a measurement of the amplitude of an analog waveform at an instant in time. The sampling rate specifies how many measurements of the analog signal amplitude are made per second by the Analog to Digital Converter (ADC) hardware of the computer. Similarly, when digital waveforms are converted to analog signals (using a Digital to Analog Converter or DAC), sampling rate refers to how many times per second the (DAC) hardware must be updated with a new sample value. At the most basic level of explanation: Analogue Sound is a wave that goes up and down (or left & right). A machine senses the wave as a change of air pressure and tries to record it. If the wave goes up & down: when the wave is up (or high) the computer will record it as a 1, when the wave goes down (or low) the computer will record it as a 0. (if the wave is goes up and stays up for a while the computer will record it as a chain of 1s). this has then converted an analogue sound wave into a digital sequence of 1s & 0s. AUDIO FILE FORMAT Is a file format for storing digital audio data on a computer system. This data can be stored uncompressed, or compressed to reduce the file size. It can be a raw bitstream, but it is usually a container format or an audio data format with defined storage layer. There are a number of different types of Audio files. The most common are Wave files (wav) and MPEG Layer-3 files (mp3). There are, however, many other audio file types discussed below. The type is usually determined by the file extension (what comes after the "." in the file name). For example, ".wav", ".mp3" or ".dct". The way the audio is compressed and stored is call the codec which determines how small the file size is. Some file types always use a particular codec. For example, ".mp3" files always use the "MPEG Layer-3" codec. Other files like ".wav" and ".dct" files support selectable codecs. For example, a ".wav" file can be encoded with the "PCM", "GSM6.10", "MPEG3" and many other codecs. Be careful not to confuse the file type with the codec - it often surprises people to know you can have a "MPEG Layer-3" encoded ".wav" file. Uncompressed audio formats, such as WAV, AIFF, AU or raw header-less PCM; Formats with lossless compression, such as FLAC, Monkey's Audio (filename extension APE), WavPack (filename extension WV), Shorten, TTA, ATRAC Advanced Lossless, Apple Lossless (filename extension m4a), MPEG-4 SLS, MPEG-4 ALS, MPEG-4 DST, Windows Media Audio Lossless (WMA Lossless), and SHN or Shorten. Formats with lossy compression, such as MP3, Vorbis, Musepack, AAC, ATRAC and Windows Media Audio Lossy (WMA lossy).