103-3 Digital Audio Introduction PDF

**103-3 -- Digital Audio Introduction** 1. Digital Audio a. Analog -- Digital Conversion - Attributes of an audio file: - File format. - Data rate. - Channels (mono, stereo, or multi-channel). - Sample rate quality. - Sample is a single measurement o...

**103-3 -- Digital Audio Introduction** 1. Digital Audio a. Analog -- Digital Conversion - Attributes of an audio file: - File format. - Data rate. - Channels (mono, stereo, or multi-channel). - Sample rate quality. - Sample is a single measurement of amplitude of a continuous analog signal - Aka a snapshot - Reading of an instance of audio - 1 sample is analogous to 1 frame of a video - Sample rate is the number of samples or measurements of amplitude, taken per second of an analog signal -- also referred to as sample frequency - CD Audio has a sample rate of 44.1 kHz -- so the A/D converter takes 44,100 measurements of amplitude per second - **Nyquist Theorem** states for desired frequency bandwidth to be faithfully encoded in the digital domain - Needs to be at least double the highest frequency - If you record at a sample rate of 20 kHz, NF would be only 10 kHz - 1 Sin wave requires 2 kHz (one for peak, one for valley) - 2 sample per frame - Keep sample rate the same across project files - **Higher sample rate** = higher fidelity, better quality, bigger file size, more CPU load - **Lower sample rate** = lower quality, smaller file sizes - **Doesn't always mean lower quality, if the audio is low hz** - If signal is above the Nyquist Frequency, over half of the NF, then aliasing occurs - Avoid aliasing -- the aliasing will mix with the actual musical frequencies, yielding an obnoxious distortion - Anti-aliasing filters get rid of the problematic frequencies (frequencies above half of the sample rate) - Oversampling is a process of sampling a signal many times higher than the selected sample rate and Nyquist frequency. - Thus, you only need one anti-aliasing filter - Sample at the highest possible frequency so that all lower frequencies are accounted for too without aliasing - Bit depth quality. - Compressed or uncompressed format. - Uncompressed Digital Audio Files - CD, WAV, AIFF - Compressed Digital Audio Files - MP3, FLAC, Dolby AC3, Dolby AC3 (DVD Audio) - Audio quality is defined by: - Sample rate. - Word length. - Whether or not the file is compressed by an audio codec (eg MP3) and what the particular codec settings are. - Extensions: -.wav - PC interchange file format. -.aiff -audio interchange file format for all computers (used by Apple). -.mp3 - popular compressed audio file format. -.mov - QuickTime can playback a wide range of compressed and uncompressed audio formats. -.sdII - DigiDesign Sound Designer II, 2-track audio editor software. - Audio file extensions only provide some info about the audio file and are simply "envelopes" that contain data - They don't provide info about the specific attributes of the audio file - There can be an.mp3 higher quality than a WAV technically   2\. Sampling Rate - Sample rate is simply the number of samples (or measurements of amplitude) taken per second. - Sample rate is also known as Sample frequency. CD quality sample rate (for example) is expressed as \"44.1KHz\", meaning simply that the converter takes 44,100 measurements of amplitude per second. Sample frequency is independent of the frequency(s) of the soundwaves being converted. - Sample bandwidth. - DO NOT confuse \"Sample Frequency\" with \"Audio Frequency\". Sample frequency is independent of the frequency(s) of the soundwaves being converted. - Higher sample rates produce better quality recordings but also bigger file sizes which demand greater space on storage devices (such as hard drives), and faster processors (CPUs) to manipulate. - Lower sample rates produce poorer quality but also smaller file sizes which demand less of storage systems, CPUs and will transfer over networks (internet) faster. - A screen shot of a computer Description automatically generated 2. **Bit Depth\ ** - Bit Depth = bits of information available for each sample - In digital audio using pulse-code modulation (PCM), bit depth is the number of bits of information in each sample, and it directly corresponds to the resolution of each sample - Pulse-code modulation (PCM) is used to digitally represent sampled analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled regularly at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. - Variations in bit depth primarily affect the noise level from quantization error---thus the signal-to-noise ratio (SNR) and dynamic range. - Techniques such as dithering, noise shaping and oversampling mitigate these effects without changing the bit depth. Bit depth also affects bit rate and file size. - Examples of Bit Depths - CD = 16 bits per sample - DVD + BluRay = 24 bits per sample - Most DAW's support 16, 24, or 32 bit audio - Bit depth sets the decibel of dynamic range - Theyre not an exact replication of the analog waveform and therefore contain quantization errors - Higher bit depth is better for recording because it sets the clipping point - One thing to note: - Consumers listen at lower bit depths - This ends up truncating the resolution and can re-introduce quantization noise - Waves get quantized incorrectly (not the same quantize as grid quantize) **Dithering:** Adds low level noise to get rid of quantization errors, getting rid of harmonic distortion - Dithering adds in white noise at a low level to fill in the blanks and make the "stair step" effect less drastic - We no longer have peaks in the noise floor because of harmonic distortion - We now have a white noise floor instead - The point of dithering is to reduce truncation when lowering bit depth \* Bit Crush is lowering bit depth used alongside a Limiter 4\. Word Clock - In order to get devices with different sample rates to be unified, have a master word clock. - AKA Primary and Secondary (instead of Master and Slave) - Word clock is what tells a A/D or D/A converter when to take the sample. It fires an impulse to the converter 44.1k (or 48k, or 96k, etc) times per second. The reliability of this clock, how evenly spaced those pulses are, determines the accuracy of the conversion process. - Jitter is manifested in audio as phase distortions and discrepancies, particularly in the higher frequencies. - Improving the quality of your clock is an easy way to upgrade every A/D and D/A converter in your studio. In most cases, adding a high-quality master clock, such as Apogee\'s Big Ben or Antelope Audio\'s Isochrone OCX will result in a marked improvement in the overall quality and clarity of your finished product. **Word clock cables:** **AES**-- 3 pin -- looks like XLR but XLR mic cable is 75 ohms, and AES/EBU is 110 ohms **S/PDIF** (Sony / Philips Digital Interface) Coaxial cable with RCA connectors -- 75 ohms **TDIF** Tascam Digital Interface **ADAT** -sending information over light **BNC** -- Coaxial cable connection 75 ohms (same as cable or broadband connection) \- it needs to be terminated (plugged) at the end or else there will be sound issues \- Can use a word clock distribution device and daisy chain 5\. Recording Project (basic components) - **Computer**: focus on getting a computer that has as much ram as possible, also getting a solid-state drive to store your DAW's and any data you want to be read as fast as possible. Store the rest of your data on a more general hard drive (1TB minimum) or an external hard drive. - **DAW**: most important part of choosing a DAW is finding out which interface appeals to your style of creativity or work flow the best. - **Audio Interface**: Most DAW's work with most brands of audio interfaces at this point, this leaves us with an endless spree of options to choose from: - Focusrite Scarlett 2i1 - Universal Audio Apollo Twin - Mackie Onyx Blackjack - M-Audio M-Track Plus 2 - Presonus AudioBox - Apogee ONE - **Studio Microphone**: Having at least one quality microphone as a creator is a necessity, this will allow you to record vocals, instruments, and possibly throw you into situations like recording podcasts or editing audio. (at least \$100) - **Headphone & Studio Monitors**: Headphones are a more important beginning stage, since you'll be able to test your music on speakers in situations such as other people\'s systems. But you need a way to at least start getting used to the way a mix sounds or feels in a certain pair of cans or speakers.   6\. Digital audio documents - There will be a save file for the actual session you work on that will store all the information and data - DAW File Management is very important to understand for each DAW you use - It is crucial for Pro Tools to always keep the session file in its parent folder, so that Pro Tools can find the samples - Pro Tools also includes in the folder all the audio files, clip groupings, session file backups and video files. - When you open up your session it's crucial to open the session file in the folder, so it recognizes all your audio files. - Digital audio documents could also include things like documents being converted to speech through narration like windows.   7\. Audio File manipulation - Optimising is the process of preparing a source/\"raw\" media file (typically video or audio) for playback on a target system, and usually involves converting it into a smaller (and usually lower quality) format. Optimising is normally required for 2 primary reasons: - To reduce a file size so that it will take up less space on a storage device. - To reduce its file size/data rate so that it will transfer faster over a network such as the internet. - Transcoding is the digital to digital conversion of one file format to another without involving an optimising (compression etc) process. Transcoding is often carried out in order to convert a file from an obsolete format to a current one. Transcoding may incur a quality loss. - (One encoding to another) - 3 types: - Lossy to lossy (AAC to MP3) - Lossless to Lossless (ALAC to FLAC) - Lossless to Lossy (FLAC to MP3) - Codec is short for compression-decompression algorithm. It is a piece of software which is employed to compress (reduce in size) a media file and then at a later time decompress it to allow it to be played or viewed. - Device or program that compressed a media file, and then uncompresses it so that it can be used - There are Hardware Codecs and Software Codecs - A codec may be lossy or lossless. - Lossless is a process whereby clever algorithms are employed to \"compact\" the data in a file without losing any information. Lossless compression can reduce the file size but not as much as lossy. - Lossy find ways to throw away information which is considered \"less-essential\". File size reduction can be dramatic (10 times smaller) but the effect on quality is usually noticeable.   Importing to ProTools -- **Add** makes it reference files from file location **Copy --** makes a new copy in the ProTools folder so that it stays within parent folder **Convert --** if you are trying to import a non-eligible file type 8\. DAW voices - Pro Tools, Reason and Logic all have a limit to the amount of voices/tracks they're able to use and have function simultaneously. -   9\. Non destructive editing - Destructive means any trimming of start and end, and any effects will process the file and once that file is saved you can\'t go back and change anything. - Non-destructive means you can go back and change it - start and end trimming will instead create cue points so to speak, effects will not be applied to the file and rendered rather than just \"remember what effect was used where and by how much.   10\. Importing/Exporting audio files - In most DAW's you have the option to import Audio,Video and MIDI information. Sometimes this may even include text formats or graphics. - You also have the option to export the same Audio, video, and MIDI information. Sometimes with programs such as Logic you can also export the Score. - Reason: - Import Audio File - Import MIDI File - Export MIDI File - Export Song as Audio File - Export Loop as Audio File - Bounce Mixer Channels - Logic: - Import: - Logic Projects - Audio File - MIDI File - Final Cut Pro XML - AAF - Other - Export: - Selection as MIDI File - Track as Audio File - All Tracks as Audio Files - Project as AAf File - Project to Final Cut Pro/XML - Score as MusicXML