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High-resolution audio (high-definition audio or HD audio) is a term for audio files with greater than 44.1 kHz sample rate or higher than 16-bit audio bit depth.It commonly refers to 96 or 192 kHz sample rates.
Xiph.org is home to other free compression formats such as Vorbis, Theora, Speex and Opus. [5] [6] [7] Version 1.3.0 was released on 26 May 2013, at which point development was moved to the Xiph.org git repository. [8] In 2019, FLAC was proposed as an IETF standard. [9] In December 2024, FLAC was formally specified in and published as RFC 9639 ...
Possible bitrate and latency combinations compared with other audio formats. Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s (or up to 256 kbit/s per channel for multi-channel tracks), frame sizes from 2.5 ms to 60 ms, and five sampling rates from 8 kHz (with 4 kHz bandwidth) to 48 kHz (with 20 kHz bandwidth, the human hearing range).
The 'Music' category is merely a guideline on commercialized uses of a particular format, not a technical assessment of its capabilities. For example, MP3 and AAC dominate the personal audio market in terms of market share, though many other formats are comparably well suited to fill this role from a purely technical standpoint.
According to Apple, audio files compressed with its lossless codec will use up "about half the storage space" that the uncompressed data would require. Testers using a selection of music have found that compressed files are about 40% to 60% the size of the originals depending on the kind of music, which is similar to other lossless formats. [3] [4]
Vorbis nominal bitrate at quality levels for 44.1 kHz stereo input. The new libvorbis v1.2 usually compresses better than these values (effective bitrate may vary). Quality Nominal bitrate Official Xiph.Org Foundation Vorbis -q-1 45 kbit/s 48 kbit/s -q0 64 kbit/s -q1 80 kbit/s -q2 96 kbit/s -q3 112 kbit/s -q4 128 kbit/s -q5 160 kbit/s -q6
A file format for the Free Lossless Audio Codec, an open-source lossless compression codec. .gsm: Designed for telephony use in Europe, GSM is used to store telephone voice messages and conversations. With a bitrate of 13 kbit/s, GSM files can compress and encode audio at telephone quality. [7] Note that WAV files can also be encoded with the ...
A PCM signal is a sequence of digital audio samples containing the data providing the necessary information to reconstruct the original analog signal.Each sample represents the amplitude of the signal at a specific point in time, and the samples are uniformly spaced in time.
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