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|Type of format||Audio codec|
The original aptX algorithm was developed in the 1980s by Dr. Stephen Smyth as part of his Ph.D research at Queen's University Belfast School of Electronics, Electrical Engineering and Computer Science; its design is based on time domain ADPCM principles without psychoacoustic auditory masking techniques.
aptX audio coding was first introduced to the commercial market as a semiconductor product, a custom programmed DSP integrated circuit with part name APTX100ED, which was initially adopted by broadcast automation equipment manufacturers who required a means to store CD-quality audio on a computer hard disk drive for automatic playout during a radio show, for example, hence replacing the task of the disc jockey.
Since its commercial introduction in the early 1990s, the range of aptX algorithms for real-time audio data compression has continued to expand with intellectual property becoming available in the form of software, firmware and programmable hardware for professional audio, television and radio broadcast, and consumer electronics, especially applications in wireless audio, low latency wireless audio for gaming and video, and audio over IP. In addition, the aptX codec can be used instead of SBC, the sub-band coding scheme for lossy stereo/mono audio streaming mandated by the Bluetooth SIG for the Advanced Audio Distribution Profile (A2DP) of Bluetooth, the short-range wireless personal-area network standard. aptX is supported in high-performance Bluetooth peripherals.
Today, both standard aptX and Enhanced aptX (E-aptX) are used in both ISDN and IP audio codec hardware from numerous broadcast equipment makers, including APT WorldCast Systems, Tieline Technology, AVT, Harris Corporation, BW Broadcast, Digigram, MAYAH, Prodys, and Qbit. An addition to the aptX family in the form of aptX Live, offering up to 8:1 compression, was introduced in 2007; and aptX Lossless, a scalable, adaptive, lossless type audio codec was announced in April, 2009.
The aptX audio codec is available for consumer and automotive wireless audio applications, notably the real-time streaming of lossy stereo audio over the Bluetooth A2DP connection/pairing between a "source" device (such as a smartphone, tablet or laptop) and a "sink" accessory (namely a Bluetooth stereo speaker, headset or headphones). aptX technology must be incorporated in both transmitter and receiver to derive the sonic benefits of aptX audio coding over the default sub-band coding (SBC) mandated by the Bluetooth standard. Consumer electronics products bearing the CSR aptX logo are certified for interoperability with other Bluetooth audio products belonging to the aptX ecosystem.
Enhanced aptX provides coding at 4:1 compression ratios for professional audio broadcast applications and is suitable for AM, FM, DAB, HD Radio and 5.1. Enhanced aptX can handle up to 4 stereo pairs of AES3 audio and compress to 1 AES3 stream for transmit. Enhanced aptX supports bit-depths of 16-bit, 20-bit or 24-bit. For 16-bit audio sampled at 48 kHz, the bit-rate for E-aptX is 384 kbit/s (2-channel), 767 kbit/s (4-channel), 1024 kbit/s (5.1-channel), and 1.28 Mbit/s (5.1 channels plus stereo). Its lowest bit-rate is 60 kbit/s for 16-bit mono audio sampled at 15 kHz, offering 7.5 kHz frequency response just below that of wideband telephony codecs (which usually operate at 16 kHz sampling rate).
aptX Live is a low-complexity audio codec that is specifically designed to maximise digital wireless microphone channel density in bandwidth-constrained scenarios, such as live performance (a.k.a. Programme Making and Special Events), where the spectrum-efficiency of radio-based devices (wireless microphones, in-ear monitoring, talk-back systems) is becoming a prime operational consideration. aptX Live offers up to 8:1 compression of 24-bit resolution digital audio streams while maintaining acoustic integrity (approx. 120 dB dynamic range) and ensuring latency of around 1.8 ms at 48 kHz sampling rates. In addition, aptX Live also features techniques that aid connection in situations where the bit error ratio (BER) is excessively high.
aptX Lossless supports high-definition audio up to 96 kHz sampling rates and sample resolutions up to 24 bits. The codec optionally permits a "hybrid" coding scheme for applications where average and/or peak compressed data rates must be capped at a constrained level. This involves the dynamic application of a form of "near lossless" coding – but only for those short sections of audio where completely lossless coding cannot respect the bandwidth constraints. Even for those short periods while the "near lossless" coding is active, high-definition audio quality is maintained, retaining audio frequencies up to 20 kHz and a dynamic range of at least 120 dB. The format is not lossless, despite the name "aptX Lossless" suggesting that it is lossless.
Coding latency is another scalable parameter within aptX Lossless and can be dynamically traded against other parameters such as levels of compression and computational complexity. The latency of the aptX Lossless codec can be scaled to as low as 1 ms for 48 kHz sampled audio, depending on the settings of other configurable parameters. aptX Lossless performs particularly well against other lossless codecs when the coding latency is constrained to be small, such as 5 ms or less, making it particularly appropriate for delay-sensitive interactive audio applications.
Many lossless codecs possess a low computational overhead compared to well-known lossy codecs, such as MP3 and AAC. This is particularly important for deeply embedded audio applications running on low-power mobile devices. aptX Lossless promotes low computational overhead by dynamically selecting the simplest coding functions for each short segment of audio whilst complying with other operational constraints, such as levels of compression and coding delay. Depending on the settings of other scalable parameters, aptX Lossless can encode a 48 kHz 16-bit stereo audio stream using only 10 MIPS on a modern RISC processor with signal processing extensions. The corresponding decoder represents only 6 MIPS on the same platform.
User metadata and special synchronization data can be incorporated into the compressed format at configurable rates. The latter permits rapid decoder resynchronization in the event of data corruption or loss over communications links where Quality of Service (QoS) can vary rapidly. Depending on the settings of parameters, decoder resynchronization can occur within 1–2 ms.
aptX Low LatencyEdit
aptX Low Latency is intended for video and gaming applications requiring comfortable audio-video synchronization whenever the stereo audio is transmitted over short-range radio to the listener(s) using the Bluetooth A2DP audio profile standard. CSR claims that aptX Low Latency for Bluetooth offers an end-to-end latency of 32 ms. The latency of standard Bluetooth stereo varies greatly depending on the system implementation and buffering. Solutions are available that use standard SBC encoding/decoding that achieve end-to-end latency of less than 40 ms. The recommended latency for Audio to video synchronization in broadcast television is within +40 ms and −60 ms (audio before/after video, respectively).
Mode of operationEdit
The example CD-quality 16-bit 44 kHz (up to 22 kHz signal bandwidth) stream is divided by two layers of 64-tap QMF (Quadrature mirror filter) into four 16-bit subbands of 11 kHz (up to 5.5 kHz signal bandwidth each). The first 64-tap QMF divides into two bands (0–11 kHz and 11–22 kHz bands), and then each one is fed into another 64-tap QMF dividing into four bands: 0–5.5 kHz, 5.5–11 kHz, 11–16.5 kHz and 16.5–22 kHz. Reduced variance is generally expected to be found in higher bands compared to lower bands, thus ADPCM is employed to allocate bits optimally.
Each band is coded with ADPCM using bit allocation of: 8 bits for band 1 (0-5.5 kHz spectrum), 4 bits for band 2 (5.5–11 kHz), 2 bits each for bands 3 and 4 (11-16.5 kHz and 16.5–22 kHz). A future modification is considered with adaptive bit reallocation based on variance analysis of each subband, for example 9,2,3,2 etc.
As a result, for mono channel, 16 bits @ 44 kHz=705 kbit/s input is converted into (4×16) 64 bits@11 kHz=705 kbit/s and then to (8+4+2+2) 16 bits @ 11 kHz=176 kbit/s.
Optionally (adds a small delay) a short-term RMSE analyzer is used to reduce dynamic range, and thus allocate bits more effectively during quiet passages (i.e. lossy format).
For a stereo signal, a standard PCM 1.4-Mbit stream is converted into 352 Kbit/sec aptX stream.
Details can be found in the EP0398973B1 patent. The main reasoning is that signal variation is reduced at higher frequencies, which makes it amenable to coding with codecs like ADPCM.
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