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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
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.
The Apple Lossless Audio Codec (ALAC, / ə ˈ l æ k /), also known as Apple Lossless, or Apple Lossless Encoder (ALE), is an audio coding format, and its reference audio codec implementation, developed by Apple Inc. for lossless data compression of digital music.
The MPEG-4 audio coding algorithm family spans the range from low bit rate speech encoding (down to 2 kbit/s) to high-quality audio coding (at 64 kbit/s per channel and higher). AAC offers sampling frequencies between 8 kHz and 96 kHz and any number of channels between 1 and 48.
FhG MP3 encoder from Adobe Audition 1.0 VBR quality 40, "Current - Best" codec. Apple iTunes 4.2 MP3 112 kbit/s VBR, Highest quality, joint stereo, smart encoding; GOGO-no-coda 3.12-b 128 -a -q 0; Audioactive Encoder 2.04 128 kbit/s High Quality; Xing MP3 Encoder 1.5 VBR quality normal; Various 12 11-22 LAME
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).
Transparency, like sound or video quality, is subjective. It depends most on the listener's familiarity with digital artifacts, their awareness that artifacts may in fact be present, and to a lesser extent, the compression method, bit rate used, input characteristics, and the listening/viewing conditions and equipment. Despite this, sometimes ...
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding.AMR is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality [3] speech starting at 7.4 kbit/s.