Tuesday, July 25, 2017

Anchoring: Digital recording of voice and voice editing



Digital Recording

In a digital recording system, sound is stored and manipulated as a stream of discrete numbers, each number representing the air pressure at a particular time. The numbers are generated by a microphone connected to a circuit called an ANALOG TO DIGITAL CONVERTER, or ADC.

digital voice recorder is a handheld device designed to record voice and sound with superior sound recording and playback, without the need for media.

In recent years, audio recording has become a bit more standardized, and a bit more restricted, so that it is pretty much platform agnostic. This page will address some basics that will hopefully cover most situations.
For digital recording, a range of devices are used, from small Sony for placement testing to a Zoom for test prompt recording, to Peak 7 on a Mac Pro for test construction. Standard go-to device is an Olympus LS-10 and Audacity for editing. Smartphones are pretty good as well, and even feature phones can do some basic recording, even if the quality is not too good.
In addition to making recordings of yourself and other people, it is often useful to have audio clips from local or online media. This used to be quite easy to do, at least in Windows XP, but it is now much more difficult to record audio that is playing on your computer into an application running on that same computer. It is not impossible, but Microsoft and Apple have pretty much cut out every way that we used to be able to do it.
However, while you won’t be able to get perfect digital copies of audio, there are ways to get recordings that are good enough. The key is having a second device to do the recording and a connector to go from the output of one to the input of the other. The tricky part is adjusting the output of the source so that it does not overwhelm the input of the recording device. This is quite difficult, and sometimes impossible, if the input level of the recorder is too high and cannot be adjusted. It also helps if you can somehow monitor the source, either directly or via some output of the recording device.

Settings for recording:

Many devices actually have settings, if you look for them, and it often helps if you adjust them according to your needs. Keep in mind that the default settings for most recording software are designed with music in mind (with a sampling rate of up to about 44,000 times per second). However, if you are just recording voice sounds, all you really need is a sampling rate of up to about 12-20,000 times per second. You can adjust the compression ratio and other settings to fit your file storage needs. Put simply, as you compress the encoding the file you save will be smaller, but the sound quality will be worse. Make sure you are using settings that fit your needs – in other words, try to use a recording and see if it has sufficient quality.
That said, one of the most important things in making recordings to share with others is NOT to make the file too large. Using the default settings of most software, you will end up with a file that is several megabytes, even though it is only one or two minutes long. In addition to the sampling rate, make sure that your settings are for mono if you are not recording in stereo, and make sure that your bitrate is as low as possible. For example, recording on my laptop, the microphone jack is not stereo, so I have no choice but to make mono recordings. Then I set the sampling rate for 11025 if I am doing a voice file. Also, using Audacity to make .mp3 files, when I save the file, I have a choice of whether to make it a constant or variable bit rate. I generally choose the constant bit rate and drop it down to 32kbs or less (most CD quality sound is 128kbs). With those settings, two minutes of sound is less than a megabyte, usually around 400 or 500KB. But again, the key is to TEST the sounds you make to see if they come out the right size and quality.
Whatever operating system you use, you should make sure that the microphone you intend to use is actually the one connected to the recording software. Often there is an integrated mic with the built-in webcam on a laptop, and this is the default microphone. If you want to use a studio-quality USB mic or headset, make that microphone is clicked as the input, and adjust the input level so that you are getting a decent level. Note that the input level for the microphone is NOT the same as the volume of the speaker on the device. These settings are usually somewhat unfamiliar to most users, so you may need to look around your control panel / preferences to find out where they are.

Notes on editing:

Depending on the audio recorder you use, and the cables that you use to connect it, and, well, the computer you use for exporting and editing, you may have a stereo file. If this is the case, then when you look at the file in Audacity, you will see two tracks. Stereo is great for music and higher quality voiceover for video, but for most purposes, you only need one channel. In Audacity, you can get rid of one channel (you want the louder of the the two) by selecting Split to Mono in the waveform interface. Other software does this in other ways, but once you split the tracks,you can just delete the one you want. This will cut your file size down considerably without any real reduction in quality.

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