How can the following information limit the recording of sound?
DSP – Digital Signal Processor
Software plug-ins or DSP emulations of multiband compressors can be complex, with many bands, and require corresponding computing power.
RAM – Random Access Memory
While the maximum RAM limit for 32-bit Windows 7 editions is 4GB, when it comes to the 64-bit editions, the amount of memory that the OS can address depends on which edition you are running.
Here are the upper RAM limits for the different editions of Windows 7:
Starter: 8GB
Home Basic: 8GB
Home Premium: 16GB
Professional: 192GB
Enterprise: 192GB
Ultimate: 192GB
These limits are similar to those for Vista editions, expect that Vista Enterprise and Vista Ultimate have had their upper limits raised from 128GB to 192GB.
File format (eg Mp3, Wav, Aiff)
MP3 is an audio coding format for digital audio which uses a form of lossy data compression. It is a common audio format for consumer audio streaming or storage, as well as a de facto standard of digital audio compression for the transfer and playback of music on most digital audio players.
The WAV format is limited to files that are less than 4 GB, because of its use of a 32-bit unsigned integer to record the file size header (some programs limit the file size to 2 GB). Although this is equivalent to about 6.8 hours of CD-quality audio (44.1 kHz, 16-bit stereo), it is sometimes necessary to exceed this limit, especially when greater sampling rates, bit resolutions or channel count are required. The W64 format was therefore created for use in Sound Forge. Its 64-bit header allows for much longer recording times. The RF64 format specified by the European Broadcasting Union has also been created to solve this problem.
Unlike the better-known lossy MP3 format, AIFF is uncompressed (which aids rapid streaming of multiple audio files from disk to the application), and is lossless. Like any uncompressed, lossless format, it uses much more disk space than MP3—about 10MB for one minute of stereo audio at a sample rate of 44.1 kHz and a bit depth of 16 bits. In addition to audio data, AIFF can include loop point data and the musical note of a sample, for use by hardware samplers and musical applications.
Audio output (eg Mono, Stereo, Surround)
Mono simply indicates the use of a single channel. Mono includes the use of a single microphone used to record sound, which is then played through a single channel through a speaker. The easiest way to check if a sound is a mono recording is through a set of headphones, incidentally, you can easily distinguish whether or not the sound plays through one headphone and not the other. Mono recording was typically used before the development of stereo recording. All the sound signals recorded are channel through a single audio channel, therefore mono sound delivers no implication of sound perspective, for instance there is no hint of the direction in which the sound is being recorded from.
Whereas mono has one independent audio channel, stereo has two. Signals that are reproduced through stereo recording have an exact correlation with each other, so when the sound is played back through either speakers or headphones, the sound is a mirrored representation of the original recording. Stereo recording would be useful in situations that require the use of sound perspective, for instance the clear location in instruments on a stage. The stereo system must have an equal cover over the two audio channels.
Why do we need 5.1? What's wrong with good old fashioned stereo? One of the fundamental limitations of stereo is that all the reverberation of the original acoustic space, whether recorded naturally or added artificially afterwards, comes out of the speakers from the front, together with the music. In real life, reverberation comes from all directions (unless you're in the open air!) and this limits how realistic stereo can ever be. 5.1, with its extra rear channels placed behind the listener, can re-create the original acoustic space more accurately than stereo ever can.
PCM – Pulse Code Modulation
There are potential sources of impairment implicit in any PCM system:
- Choosing a discrete value that is near but not exactly at the analog signal level for each sample leads to quantization error.
- Between samples no measurement of the signal is made; the sampling theorem guarantees non-ambiguous representation and recovery of the signal only if it has no energy at frequency fs/2 or higher (one half the sampling frequency, known as the Nyquist frequency); higher frequencies will generally not be correctly represented or recovered.
- As samples are dependent on time, an accurate clock is required for accurate reproduction. If either the encoding or decoding clock is not stable, its frequency drift will directly affect the output quality of the device.
In what types of scenario may you use the following audio recording equipment?
Multi-track recording
Midi – Multi
Instrument Interface
DAT, Analogue
Software Plug-in’s
Software Sequencer