Audio data is defined by certain parameters, which determine its quality and format. The major ones are: the sampling rate, number of bits per sample, and the total number of channels. However, before we take a look at these, let’s first understand what they’re all about.

Digital audio files are created by taking an analog audio stream and sampling it at frequent intervals. Sampling rates are specified in kHz and are measured per channel. So audio data recorded in stereo at 16,000 samples/sec is actually 32,000 samples/sec, since stereo recording involves dual channels. A higher sampling rate means more number of samples captured per second, and hence better quality sound. For normal listening, the playback of recorded sound at a slightly different rate doesn’t show significant difference in hearing, but for professional musicians who do real-time editing of audio, even slight differences matter.

The number of bits per sample defines the size of the sampled data packets. This can be either 8 or 16 bits. Coming to channels, the number of channels can be equated to the number of paths over which the complete sound is split. This can either be on the basis of instruments or frequencies. In professional music recordings, the sound for different instruments is recorded on to different paths or tracks. This makes them easier to edit. For playback, the hardware required to handle, say 16 channels, will turn out to be very expensive. So stereo sound or a maximum of 5+1 channel Dolby surround is enough for homes. Some movie theatres, however, do have a setup for giving you true 16-channel surround sound.

Making waves

The WAV or ‘wave’ format can be considered as the basic uncompressed form of audio. Almost all primary recordings of audio or music are done in WAV format. Since it’s uncompressed, all data remains intact, making it easy to edit. A WAV file consists of uncompressed 8 or 16-bit sound samples with a header preface and specifications on how the audio data is formatted in the file. The sample rates used are typically 11,025 sample/sec for telephone quality, 22,050 sample/sec for radio quality or 44,100 sample/sec for CD quality. As for the number of channels, hold your breath, it’s possible to encode up to 65,536 channels. Most other audio formats use some compression algorithm or the other, which reduces their size making them unsuitable for editing.

MIDI

Moving on from WAV, another old favorite in audio is the MIDI format. MIDI stands for Musical Instrument Digital Interface. It’s basically a standard for storing sounds from musical instruments like synthesizers and keyboards. This format doesn’t have any actual sound data in it. Typically, an MIDI file has instructions that include the note’s pitch, length and volume. Plus, there are a few other parameters like envelopes, rise, attack and delay for sound that can also be included in MIDI instructions.

These instructions then direct the digital-to-analog converter in your sound card to equate them to actual sound samples from the sound bank for the particular sound card. So, though not always, there’s a possibility of hearing the same MIDI files differently with two different sound cards. For professional musicians, this music format is a must as it’s easy to edit and has lower levels of noise. Most good quality synthesizers in the market have a MIDI interface that can be connected to PCs and used to create scores. Software is also available that can convert MIDI sample to written music notes. So you can simply print these notes and use them for your music class.