<Mozilla>: Even modest quality, high-fidelity | stereo sound can use a substantial amount of disk space. For web developers, an even bigger concern is the network bandwidth needed in order to transfer audio, whether for streaming or to download it for use during gameplay. The processing of audio data to encode and decode it is handled by an audio codec (COder/DECoder).
H3S1: AAC (Advanced Audio Coding):
<Mozilla>: The Advanced Audio Coding (AAC) codec is defined as part of the MPEG-4 | (H.264) standard; specifically, as part of MPEG-4 Part 3 and MPEG-2 Part 7. Designed to be able to provide more | compression with higher audio fidelity than MP3, AAC has become a popular choice, and is the standard | format for audio in many types of media, including Blu-Ray | discs and HDTV, as well as being the format used for songs purchased from online vendors including iTunes.
H3S2: MP3 (MPEG-1 Audio Layer III):
<Mozilla>: Of the audio | formats specified by the MPEG/MPEG-2 standards, MPEG-1 Audio Layer III—otherwise known as MP3—is by far the most widely used and best-known. The MP3 | codec is defined by MPEG-1 Part 3 and MPEG-2 Part 3, and was introduced in 1991 (and finalized in 1992).
When MP3-format audio is stored inside an MPEG container, the resulting file is also referred to as just an “MP3 file” or “MP3.” Files with the ubiquitous
.mp3 extension are stored in what is perhaps the most widely distributed audio file format in the world, which is in large part responsible for the digital audio revolution of the late 1990s and early 2000s.
MPEG-1 MP3 audio supports higher bit rates as well as higher sampling rates than MP3 audio in MPEG-2 files. MPEG-1 format MP3 is generally best for music or other complex audio, while MPEG-2 mode MP3 audio is acceptable for speech and other simpler sounds.
The patents behind MP3 have expired, removing many or most licensing concerns around using MP3 files in your projects. That makes them a good choice for many projects.
|Supported bit rates||MPEG-1 mode: 32 kbps, 40 kbps, 48 kbps, 56 kbps, 64 kbps, 80 kbps, 96 kbps, 112 kbps, 128 kbps, 160 kbps, 192 kbps, 224 kbps, 256 kbps, 320 kbps|
MPEG-2 mode: 8 kbps, 16 kbps, 24 kbps, 32 kbps, 40 kbps, 48 kbps, 56 kbps, 64 kbps, 80 kbps, 96 kbps, 112 kbps, 128 kbps, 144 kbps, 160 kbps
|Variable Bit Rate (VBR) support||Yes|
|Supported sample formats||16-bit integer|
|Supported sample rates||MPEG-1 mode: 32000 Hz, 44100 Hz, 48000 Hz|
MPEG-2 mode: 16000 Hz, 22050 Hz, 24000 Hz (Half the frequency of the MPEG-1 supported modes)
|Recommended minimum bit rate for stereo sound||128 kbps at 48 kHz sample rate|
|Maximum audio channels||MPEG-1 mode: 2 [2.0]|
MPEG-2 mode: 5 (plus 1 optional Low Frequency Enhancement channel) [5.1]
|Audio frequency bandwidth||Varies, depending on bit rate and psychoacoustic analysis|
|Latency||At least 100 ms|
|Browser compatibility||FeatureChromeEdgeFirefoxInternet ExplorerOperaSafariMP3 supportYesYesYes9Yes3.1|
|Container support||MPEG-1, MPEG-2, MP4, ADTS, 3GP|
|RTP / WebRTC compatible||No|
|Licensing||Patent-free in the EU as of 2012; patent-free in the United States as of April 16, 2017; now free to use|
H3S3: Encoding audio waves into binary:
H4S1: Encoding parameters:
<Mozilla, audio concepts>: The higher the amplitude (height) of the wave, the louder the sound is at that instant. The shorter the wavelength (the closer together the crests of the wave are), the higher the frequency (or pitch) of the sound that’s produced.
H4S2: Analog to digital transcoding:
<Mozilla>: Sound enters the computer through a microphone or other input in the form of a stream of electrons whose voltage | varies to represent the amplitude of the sound | wave. This analog signal is then converted into digital form by a circuit that captures the incoming wave’s amplitude at regular intervals, converting that data into a number in a form that is understood by the audio recording system. Each of these captured moments is a sample. By chaining all the samples together, you can approximately represent the original wave, as seen in the diagram below.
The more often you take samples of the original audio, the closer to the original you can get. The number of samples taken per second is called the sample rate. At the most basic level<in-computing>, audio is represented by a stream of samples, each specifying the amplitude of the audio waveform as measured for a given | slice of the overall waveform of the audio signal.
H5S1: Channels: The position of each audio source within the audio signal is called a channel. Each channel contains a sample indicating the amplitude of the audio being produced by that source at a given moment in time. For instance, in stereo sound, there are two audio sources: one speaker on the left, and one on the right. Each of these is represented by one channel, and the number of channels contained in the audio signal is called the channel count. H6S1: Frames: While recording or generating multi-channel audio files, the channels are assembled into a series of audio frames, each consisting of one sample for each of the audio’s channels. An individual sample is a numeric value representing the amplitude of the sound waveform at a single moment in time, and may be represented in various formats.
H5S2: Common values: Stereo audio is probably the most commonly used channel arrangement in web audio, and 16-bit samples are used for the majority of day-to-day audio in use today. For 16-bit stereo audio, each sample taken from the analog signal is recorded as two 16-bit integers, one for the left channel and one for the right. That means each sample requires 32 bits of memory. At the common sample rate of 48 kHz (48,000 samples per second), this means each second of audio occupies 192 kB of memory. Therefore, a typical three-minute song requires about 34.5 MB of memory. That’s a lot of storage, but worse, it’s an insane amount of network bandwidth to use for a relatively short piece of audio. That’s why most digital audio is compressed.
It is always, though not always clear, that moves in the chess openings are based on certain definite ideas. These ideas form the “background | and foundation”, while the moves themselves represent actual | construction. Reuben Fine, GM, Ph-D pg-1 <a-r> Pre-reading operation: play 10 games of any time control with a partner using the following opening sets, potentially with an option, or the loser, to change their setup before the next game, but always with at least one person using one of the setups, before moving on to read the rest of the section. |=P An alternate opening state for learning about playing with idealized setups for both sides, one for white and one for black, presented in 3 formats: all pieces, only white, only black.
1. e4 is inherently a | gambit in that it thrusts a 4x defended pawn into an undefended square, -4 effect, as one of only 2 tetra defended pawns for each side in the initial board position with the d-pawn being the other, a risk accepted in order to attain +4 mobility for the d1 queen and +5 for the f1 bishop. This analysis further reveals that gambits, at the core, are an exchange of defensive effect of the gambitted object for increased mobility for the gambitting side. |=P In comparison, 1. d4 is less of a gambit because the exchange is -3 defensive effect, as the pawn remains defended by the d1 queen, for +5 mobility of the c1 Bishop and +2 mobility of the queen, a 25% reduction in effect loss and a 22% reduction in mobility, but it’s close.
|=P e4 may be favored among younger players due the potential of developing the queen to h5 and Bishop to c4 for a siege on the f-pawn, which, like all non-central pawns in the standard start, are 1x defended, and hitting several of the thematic easy checkmates on novice friends such as the Scholar’s and Legal’s, as well as any combinations that can arise in the Fried Liver. Post-script: Alt-purpose of chess: to refute wrong ideas: Reuben Fine pg-1: But the man who knows that Black has neglected the center, deprived his KKt <ch-10 Chpt-mn H3S1 H4S1> of its best square, and weakened his King position will find it a simple matter to refute his opponent’s faulty play. The bane of a logical player’s existence is a haphazard tactical blunder that destroyed his chances to refute profound positional mistakes by a less experienced opponent, but 4th dimensional <r: ch-2 H2S5> time travel via analysis and return to tabiya # allows us to revisit those moments again and drive our theoretical refutation home. Fin.
H7S1: H2S4 H3S1: Che opening theor x math, stat [2 MM]:
An important counterpoint to the central thesis of Material in Chess, for which the thesis statement is presented in <ch-13 H2S3 H3S2>, that chess is a purely | abstract science from which all logical operators originate, is found in the field of opening theory within chess theory <fbno>, a field as ancient as chess theory itself, as the first published chess book contained both <r: ch-6 H2S1 H3S4, Lucena> Opening theory is inductive rather than deductive because we have, for the past 600 years, lacked the necessary | computing power whether biological or artificial # to deductively solve any positions from the development phase <ch-1 H2S1>, and therefore, we must use a less peak intensive #np but overall less efficient method of trial and error to simply run a large number of simulation experiments under control | and quantity #np-2 If it is favorable for White, theory concerns itself with the improvement of Black’s defensive possibilities. Conversely, if, as is usually the case, it is even, the problem is to better White’s play. Fine, GM pg-4 <a-r> WIth this statement from Ben Fine in 1943, chess establishes itself as a field of inductive science because its practitioners have demonstrated not only 1) gathering statistics <r: ch-19 H2S3 H3S4, sci-method, collective data> but also 2) drawing conclusions Fin-v1.
H7S2: H2S4 H3S2: Che opening d-bs x comp-sci, databases :
|⭐ EZ-Directive: The central thesis of chess | opening theory is that f holds an advantage | by default and therefore s’s first task in a game is to equalize before seeking an advantage, while seeking advantage before equality # is a form of attacker’s neglect ch-44 h2s3 https://lichess.org/WPHxXOLS/black#52|
This thesis can be verified by an examination of the chess opening database, but before we do this, we need to understand how databases work in the language of computer science: MIC: The first known instance of a tablebase, and perhaps database in general, was founded in millennial Arabia: Murray <a-r> <pg-27> The Muslim MSS, not only contain a number of positions illustrating the end-game, but also summarize the experience of players in tables which give the ordinary result when the players were left with the forces stated. Thus, the Rook wins against Firz, and Pawn defended by the Firz, draws against the Knight, two Fils and Pawn, Queen and Firz, two Firzes, two Knights and Firz. Some 130 cases are similarly | classified in all.
<Mozilla>: The process of compressing and decompressing audio is performed by encoding and decoding it using an audio codec (COder/DEcoder). Hu: The process of compressing and decompression is a subset of any encoding-decoding paradigm, with the idea of # saving bit.rate-in,transit<Turing>by off-loading capacity to the endpoints<Turing-2>; however, the paradigm has to be lossless in order for the data compression to not cost experiential–quality<Turing-3>
H4S4: Sample rate:
Mozilla: The most common sample rates are: H5S1: 8000 Hz: The international G.711 standard for audio used in telephony uses a sample rate of 8000 Hz (8 kHz). This is enough for human speech to be comprehensible. 44100 Hz: The 44.1 kHz sample rate is used for compact disc (CD) audio. CDs provide uncompressed 16-bit stereo sound at 44.1 kHz. Computer audio also frequently uses this frequency by default. 48000 Hz: The audio on DVD is recorded at 48 kHz. This is also often used for computer audio. 96000 Hz: High-resolution audio. 192000 Hz: Ultra-high resolution audio. Not commonly used yet, but this will change over time.
Hu: Since the sample rate is a factor of frequency, and this is because at higher frequencies, or higher pitch sounds, the amplitudes are closer | together, which means the brain needs to be working faster in order to tease them apart<Turing><#n-p>
<Mozilla>: There is a reason why 44.1 kHz is considered the minimum “high fidelity” sampling rate. The Nyquist-Shannon sampling theorem dictates that to reproduce a sound accurately, it must be sampled at twice the rate of the sound’s frequency. Since the range of human hearing is from around 20 Hz to 20,000 Hz, reproducing the highest-pitched sounds people can generally hear requires a sample rate of more than 40,000 Hz.
To provide additional room for a low-pass filter in order to avoid distortion caused by aliasing, an additional 2.05 kHz transition band is added to the pre-sampling frequency (resulting in 22,050 Hz). Doubling that per the Nyquist theorem results in a final minimum frequency of (you guessed it) 44.1 kHz.
Wikipedia: The voiced speech of a typical adult male will have a fundamental frequency from 85 to 155 Hz, and that of a typical adult female from 165 to 255 Hz. Hu: Based on the Nyquist-Shannon sampling theorem posited previously, no more than 500 Hz, or 0.5 k-Hz, is the theoretical minimum for the encoding of human speech, provided by that the rest of the acoustic environment in computing is solved<Turing>; this represents a 100:1 reduction in compression from computer audio. If only the higher.frequency-sounds would get distorted, encoding at 8000 Hz, then this is perhaps a reasonable sample rate for microphone input, of human.speech-only,input. We can use a separate encoding paradigm, with a higher sample-rate, if music is expected in the input, including from desktop–audio<anutha-one><Turing>
Peterson, Reddy, and Hamel: The auditory system processes how we hear and understand sounds within the environment. It is made up of both peripheral structures (e.g., outer, middle, and inner ear) and brain regions (cochlear nuclei, superior olivary nuclei, lateral lemniscus, inferior colliculus, medial geniculate nuclei, and auditory cortex).
Auditory information ascending through the auditory pathways start at the auditory nerve. These nerves synapse within the cochlear nucleus. A majority of auditory information is then transmitted through crossing fibers into the superior olivary complex. From there, the information ascends through the contralateral side of the brainstem and brain to the cortex.
https://www.mpeg.org/: This is the home page of MPEG<The Moving Picture Experts Group>, the group that develops standards for coded representation of digital audio, video, 3D Graphics and genomic data. Since its establishment in 1988, the group has produced standards that help industry offer end users an ever more enjoyable digital media experience. MPEG developed well known video and audio coding standards such as AVC, HEVC, VVC and MP3.
[Long!] Definition of the Opus Audio Codec: https://datatracker.ietf.org/doc/html/rfc6716 RTP Payload Format for the Opus Speech and Audio Codec: https://datatracker.ietf.org/doc/html/rfc7587 Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN): https://datatracker.ietf.org/doc/html/rfc3389 WebRTC Audio Codec and Processing Requirements: https://datatracker.ietf.org/doc/html/rfc7874
Peterson DC, Reddy V, Hamel RN. Neuroanatomy, Auditory Pathway. [Updated 2022 Aug 8]. In: StatPearls [Internet]. Treasure Island (FL): StatPearls Publishing; 2022 Jan-. Available from: https://www.ncbi.nlm.nih.gov/books/NBK532311/