Development of the MPEG-1 standard began in May 1988. Fourteen video and fourteen audio codec proposals were submitted by individual companies and institutions for evaluation. The codecs were extensively tested for computational complexity and subjective (human perceived) quality, at data rates of 1.5 Mbit/s. This specific bitrate was chosen for transmission over T-1/E-1 lines and as the approximate data rate of audio CDs.[17] The codecs that excelled in this testing were utilized as the basis for the standard and refined further, with additional features and other improvements being incorporated in the process.[18]
MPEG-1 Audio is divided into 3 layers. Each higher layer is more computationally complex, and generally more efficient at lower bitrates than the previous.[16] The layers are semi backwards compatible as higher layers reuse technologies implemented by the lower layers. A "Full" Layer II decoder can also play Layer I audio, but not Layer III audio, although not all higher level players are "full".[56]
Codec Mpeg Audio Layer 1 2 3 Mpga Youtube
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MPEG-1 Audio Layer II was derived from the MUSICAM (Masking pattern adapted Universal Subband Integrated Coding And Multiplexing) audio codec, developed by Centre commun d'études de télévision et télécommunications (CCETT), Philips, and Institut für Rundfunktechnik (IRT/CNET)[16][18][58] as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of digital audio broadcasting.
MPEG-1 Audio Layer III was derived from the Adaptive Spectral Perceptual Entropy Coding (ASPEC) codec developed by Fraunhofer as part of the EUREKA 147 pan-European inter-governmental research and development initiative for the development of digital audio broadcasting. ASPEC was adapted to fit in with the Layer II model (frame size, filter bank, FFT, etc.), to become Layer III.[18]
.mp3 is the most common extension for files containing MP3 audio (typically MPEG-1 Audio, sometimes MPEG-2 Audio). An MP3 file is typically an uncontained stream of raw audio; the conventional way to tag MP3 files is by writing data to "garbage" segments of each frame, which preserve the media information but are discarded by the player. This is similar in many respects to how raw .AAC files are tagged (but this is less supported nowadays, e.g. iTunes).
Mac and iOS only supports AAC and MP3 Audio CODEC, while my video has MPEG Audio layer 1/2 Audio CODEC. So, while converting/compressing this video, my client has selected audio CODEC to default rather than AAC.
For that reason, the codecs parameter can be added to the MIME type describing media content. With it, container-specific information can be provided. This information may include things like the profile of the video codec, the type used for the audio tracks, and so forth.
This guide briefly examines the syntax of the media type codecs parameter and how it's used with the MIME type string to provide details about the contents of audio or video media beyond indicating the container type.
All media types based upon the ISO Base Media File Format (ISO BMFF) share the same syntax for the codecs parameter. These media types include MPEG-4 (and, in fact, the QuickTime file format upon which MPEG-4 is based) as well as 3GP. Both video and audio tracks can be described using the codecs parameter with the following MIME types:
The Audio Object Type is specified as a one or two digit decimal value (unlike most other values in the codecs parameter, which use hexadecimal). For example, MPEG-4's AAC-LC has an audio object type number of 2, so the full codecs value representing AAC-LC is mp4a.40.2.
As part of a move toward a standardized and powerful format for the codecs parameter, WebM is moving toward describing video content using a syntax based on that defined by the ISO Base Media File Format. This syntax is defined in VP Codec ISO Media File Format Binding, in the section Codecs Parameter String. The audio codec continues to be indicated as either vorbis or opus.
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). In this article, we look at audio codecs used on the web to compress and decompress audio, what their capabilities and use cases are, and offer guidance when choosing audio codecs to use for your content.
Additionally, WebRTC implementations generally use a subset of these codecs for their encoding and decoding of media, and may support additional codecs as well, for optimal cross-platform support of video and audio conferencing, and to integrate better with legacy telecommunication solutions. See Codecs used by WebRTC for details.
There are two general categories of factors that affect the encoded audio which is output by an audio codec's encoder: details about the source audio's format and contents, and the codec and its configuration during the encoding process.
Audio codecs typically employ cleverly-designed and highly-complex mathematical algorithms to take source audio data and compress it to take substantially less space in memory or network bandwidth. In addition to choosing the type of encoder to use, you may have the opportunity to adjust the encoder using parameters that choose specific algorithms, tune those algorithms, and specify how many passes to apply while encoding.
There are two basic categories of audio compression. Lossless compression algorithms reduce the size of the audio without compromising the quality or fidelity of the sound. Upon decoding audio compressed with a lossless codec such as FLAC or ALAC, the result is identical in every way to the original sound, down to the bit.
Because of the differences in how lossy codecs work compared to lossless ones, especially the fact that lossless ones have to be much more conservative with their compression, lossy codecs nearly always result in significantly smaller compressed audio than lossless codecs do.
Generally speaking, the most common reasons to choose lossless audio are because you require archival-quality storage, or because the audio samples will be remixed and recompressed, and you wish to avoid the amplification of artifacts in the audio due to recompression. For real-time streaming of audio, a lossy codec is usually required in order to ensure the flow of data can keep up with the audio playback rate regardless of network performance.
In addition to providing audio for specific speakers in a sound system, some codecs may allow audio channels to be used to provide alternative audio, such as vocals in different languages or descriptive audio for visually impaired people.
The audio frequency bandwidth of a codec indicates the range of audio frequencies that can be represented using the codec. Some codecs operate specifically by eliminating audio that falls outside a given frequency range. There is a correlation between the sample rate and the maximum sound frequency that can be represented by a waveform represented by a codec. At a theoretical level, the maximum frequency a codec can represent is the sample rate divided by two; this frequency is called the Nyquist frequency. In reality, the maximum is slightly lower, but it's close.
The audio frequency bandwidth comes into play especially vividly when a codec is designed or configured to represent human speech rather than a broad range of sounds. Human speech generally resides within the audio frequency range of 300 Hz to 18 kHz. However, the vast majority of human vocalizations exist in the range 300 Hz to 8 kHz, and you can capture enough of human vocalizations in the frequency range 500 Hz to 3 kHz to still be understandable.
For that reason, speech-specific codecs often begin by dropping sound that falls outside a set range. That range is the audio frequency bandwidth. G.722, for example, strips away sounds outside the audio frequency bandwidth of 50 Hz to 7 kHz. This reduces the amount of data that needs to be encoded from the outset.
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. 2ff7e9595c
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