AptX

Last updated
aptX
QC aptX Vertical Black 4754.png
Internet media type audio/aptx
Developed by Qualcomm [1] [2]
Type of format Audio codec

aptX (apt stands for audio processing technology [3] ) is a family of proprietary audio codec compression algorithms owned by Qualcomm, with a heavy emphasis on wireless audio applications.

Contents

History

The original compression 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 [4] its design is based on time domain ADPCM principles without psychoacoustic auditory masking techniques.

The algorithm was then commercialized under the name aptX and 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.

The company was bought by Solid State Logic ca. 1988, and became part of Carlton Communications plc. [5] in 1989. In the early 1990s, APT's codecs were used to transfer audio via ISDN lines [6] [7] by companies such as Disney to check dubbing done in Europe. On 23 December 1999, [8] Solid State Logic was sold by Carlton Communications to 3i. [9] [10] Then in 2005, Solid State Logic sold APT in a management buyout. [4] [3] On 1 March 2005 APT Licensing Ltd. was incorporated in Belfast (until 18 December 2008 named Audio Processing Technology Holdings Limited). [11]

The range of aptX algorithms for real-time audio data compression expanded 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. [12] In addition, the aptX codec was introduced as an alternative to 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. [13] [14]

Today, standard aptX and Enhanced aptX (E-aptX) are used in both ISDN and IP audio codec hardware from numerous broadcast equipment makers. An addition to the aptX family in the form of aptX Live, offering up to 8:1 compression, was introduced in 2007; [15] and aptX HD, a lossy, but scalable, adaptive, "near-lossless" quality audio codec was announced in April 2009. [16]

The company split in 2009; the broadcast hardware business was acquired by Audemat and became part of what is now WorldCast Systems, and the licensing business became APT Licensing and was acquired by CSR plc in 2010. [17] aptX was previously styled apt-X until acquired by CSR in 2010. CSR was subsequently acquired by Qualcomm in August 2015. [18]

Variants

aptX variants [19]
SBC aptXaptX LLaptX HDaptX Adaptive Audio CD
(for reference)@ 279 kbit/s@ 420 kbit/s@ up to ~1.2 mbit/s(for reference)
misc.Launch?< 2009 [20] 2012 [21] 2016 [20] 2018 [20] 2021 [22] 1980s
Trademark Qualcomm
(until August 2015: CSR, until July 2010: APT Licensing Ltd., until March 2005: Solid State Logic)
Related patents EP 0400755B1   [23] (expired) EP 0398973B1   [24] (revoked)aptX,
US 9398620B1   [25] (expired)
aptX ?
Free implementations FFmpeg, libsbc FFmpeg, libopenaptx FFmpeg, libopenaptx FFmpeg, libopenaptxN/A
Proprietary implementations MultipleQualcomm libaptX [26] NoneQualcomm libaptXHD [26]  ?
ChipCSR8635CSR8670CSR8675QCC5100
Audio
Encoding
Word depth ?16-bit16-bit16-bit
24-bit
24-bit16-bit16-bit
Sampling rate 44.1 kHz
48 kHz
 
44.1 kHz
48 kHz
 
44.1 kHz
48 kHz
 
44.1 kHz
48 kHz
 
44.1 kHz
48 kHz
96 kHz
44.1 kHz44.1 kHz

 
Bit rate 345 kbit/s(@ 48 kHz)352 kbit/s(@ 44.1 kHz)
384 kbit/s(@ 48 kHz)
352 kbit/s(@ 48 kHz)576 kbit/s(24 bits @ 48 kHz)279 kbit/s420 kbit/s~140 kbit/s to 1.2 mbit/s (content dependent) [27] [22] 1411 kbit/s(@ 44.1 kHz)
ConstantConstantConstantConstantVariableVariableConstant
Codec Latency  ?1.8 – 2.0 ms ?1.8 – 2.0 ms1.4 – 2.0 ms ?
Hardware transmitter latency  ? ?≈ 40 ms [28]
(using dedicated antenna)
 ?≈ 80 ms [29]  ?
Software transmitter latency (most phones)200 – 500 ms [30]
depending on the transmitting device
Backwards compatible withSBC[ citation needed ]SBC, aptXSBC, aptXSBC, aptX, aptX HD ? ?
Sound
quality
THD+N @ 1 kHz−67 dB [31] or −85 dB? [19] −85 dB-80 dB [32] or −90 dB? [19] −90 dB−100 dB−96 dB−96 dB
Multi-tone @ 1 kHz−100 dB ?−100 dB−90 dB−100 dB ?
Multi-tone @ 10 kHz−65 dB ?−90 dB−85 dB−95 dB ?
Crosstalk −120 dB ?−155 dB−90 dB−200 dB ?
SNR @ 1 kHz93 dB93 dB129 dB130 dB135 dB−96 dB−96 dB
PEAQ ODG−0.18 [31] or −0.08? [19]  ?0.05 [32] or 0.04? [19] −0.060.045 ?
Frequency response over BT20 Hz – 22.7 kHz20 Hz – 22.7 kHz20 Hz – 22.7 kHz20 Hz – 22.7 kHz20 Hz – 22 kHz20 Hz – 22 kHz

aptX

The aptX audio codec is used 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 (e.g. a Bluetooth stereo speaker, headset or headphones). The 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. Products bearing the CSR aptX logo are certified for interoperability with each other. [33]

Enhanced aptX

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, 20 or 24 bit. For audio sampled at 48 kHz, the bit-rate for E-aptX is 384 kbit/s (dual channel), 768 kbit/s (quad channel), 1024 kbit/s (5.1-channel), and 1280 kbit/s (5.1 channels plus stereo). Its lowest bit-rate is 60(?) kbit/s for mono audio sampled at 16 kHz, offering about 7.5 kHz frequency response just below that of wideband telephony codecs (which usually operate at 16 kHz sampling rate). [34]

aptX Live

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 rate (BER) is excessively high.

aptX LL

aptX LL or 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. The technology offers an end-to-end latency of 32 ms over Bluetooth. By comparison, 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). [35] However, AptX Low Latency requires a dedicated, wireless antenna, so it did not achieve much adoption in smartphones and was retired by Qualcomm in favor of aptX Adaptive. [36] [37] Its main competitor is the LLAC.

aptX HD

aptX HD or aptX High Definition has bit-rate of 576 kbit/s. It supports high-definition audio up to 48 kHz sampling rates and sample resolutions up to 24 bits. Unlike the name suggests, the codec is still considered lossy; [38] however, it permits a "hybrid" coding scheme for applications where average or peak compressed data rates must be capped at a constrained level. This involves the dynamic application of "near lossless" coding for those sections of audio where completely lossless coding is impossible due to bandwidth constraints. "Near lossless" coding maintains a high-definition audio quality, retaining audio frequencies up to 20 kHz and a dynamic range of at least 120 dB. Its main competitors are LDAC codec developed by Sony and LHDC.

Another scalable parameter within aptX HD is coding latency. It can be dynamically traded against other parameters such as levels of compression and computational complexity. The latency of the aptX HD codec can be scaled to as low as 1 ms for 48 kHz sampled audio, depending on the settings of other configurable parameters. aptX HD 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 the benefit of 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 HD 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 HD 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. [39] [40]

aptX Adaptive

aptX Adaptive is a next-generation dynamically adjustable audio codec intended for premium audio quality and low-latency. aptX Adaptive's bitrate scales dynamically between 279 kbit/s and 420 kbit/s. It also works with a shared, rather than dedicated, wireless antenna. [36] Qualcomm claims their new compression algorithm provides a compression ratio between 5:1 to 10:1. This allows aptX Adaptive at 279 kbit/s and 420 kbit/s to produce the same sound quality as aptX at 352 kbit/s and aptX HD at 576 kbit/s. aptX Adaptive supports 16 and 24 bit-depths at 44.1, 48, and 96 kHz sample rates. Actual 96 kHz support, however, is depending on actual product hardware implementation. For example, B&O H95 only support 48 kHz at max even with AptX Adaptive support. Hardware aptX Adaptive has end-to-end latency of 80 ms, [29] but most phones are using a software transmitter - which does not have any latency advantages over other codecs. [41] aptX Adaptive is also backward compatible with older aptX and aptX HD codecs, but not with aptX LL. [36] [42] [43] [44]

aptX Voice

aptX Voice is a new feature of aptX Adaptive, and is designed to significantly improve the quality of voice for those using Bluetooth accessories to make voice calls. It does this by delivering 32 kHz voice call quality within the Bluetooth Hands-Free Profile. [45]

aptX Lossless

In 2021, Qualcomm announced aptX Lossless, described as a new capability of the aptX Adaptive codec aimed to deliver CD-quality, 16-bit, 44.1kHz lossless audio over Bluetooth. aptX Lossless utilizes Qualcomm Bluetooth High Speed Link technology to scale up to about 1.2 mbit/s in favorable RF environments. [22]

Mode of operation

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 @ 44.1 kHz=705.6 kbit/s input is converted into (4×16) 64 bits@11.025 kHz=705.6 kbit/s and then to (8+4+2+2) 16 bits @ 11.025 kHz=176.4 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 [46] patent. The main reasoning is that signal variation is reduced at higher frequencies, which makes it amenable to coding with codecs like ADPCM.

See also

Related Research Articles

A codec is a device or computer program that encodes or decodes a data stream or signal. Codec is a portmanteau of coder/decoder.

In information theory, data compression, source coding, or bit-rate reduction is the process of encoding information using fewer bits than the original representation. Any particular compression is either lossy or lossless. Lossless compression reduces bits by identifying and eliminating statistical redundancy. No information is lost in lossless compression. Lossy compression reduces bits by removing unnecessary or less important information. Typically, a device that performs data compression is referred to as an encoder, and one that performs the reversal of the process (decompression) as a decoder.

<span class="mw-page-title-main">Lossy compression</span> Data compression approach that reduces data size while discarding or changing some of it

In information technology, lossy compression or irreversible compression is the class of data compression methods that uses inexact approximations and partial data discarding to represent the content. These techniques are used to reduce data size for storing, handling, and transmitting content. The different versions of the photo of the cat on this page show how higher degrees of approximation create coarser images as more details are removed. This is opposed to lossless data compression which does not degrade the data. The amount of data reduction possible using lossy compression is much higher than using lossless techniques.

Windows Media Audio (WMA) is a series of audio codecs and their corresponding audio coding formats developed by Microsoft. It is a proprietary technology that forms part of the Windows Media framework. WMA consists of four distinct codecs. The original WMA codec, known simply as WMA, was conceived as a competitor to the popular MP3 and RealAudio codecs. WMA Pro, a newer and more advanced codec, supports multichannel and high-resolution audio. A lossless codec, WMA Lossless, compresses audio data without loss of audio fidelity. WMA Voice, targeted at voice content, applies compression using a range of low bit rates. Microsoft has also developed a digital container format called Advanced Systems Format to store audio encoded by WMA.

Adaptive Transform Acoustic Coding (ATRAC) is a family of proprietary audio compression algorithms developed by Sony. MiniDisc was the first commercial product to incorporate ATRAC, in 1992. ATRAC allowed a relatively small disc like MiniDisc to have the same running time as CD while storing audio information with minimal perceptible loss in quality. Improvements to the codec in the form of ATRAC3, ATRAC3plus, and ATRAC Advanced Lossless followed in 1999, 2002, and 2006 respectively.

Dolby Digital, originally synonymous with Dolby AC-3, is the name for a family of audio compression technologies developed by Dolby Laboratories. Called Dolby Stereo Digital until 1995, it is lossy compression. The first use of Dolby Digital was to provide digital sound in cinemas from 35 mm film prints. It has since also been used for TV broadcast, radio broadcast via satellite, digital video streaming, DVDs, Blu-ray discs and game consoles.

MPEG-4 Part 3 or MPEG-4 Audio is the third part of the ISO/IEC MPEG-4 international standard developed by Moving Picture Experts Group. It specifies audio coding methods. The first version of ISO/IEC 14496-3 was published in 1999.

In telecommunications and computing, bit rate is the number of bits that are conveyed or processed per unit of time.

<span class="mw-page-title-main">Sound quality</span> Assessment of the audio output from an electronic device

Sound quality is typically an assessment of the accuracy, fidelity, or intelligibility of audio output from an electronic device. Quality can be measured objectively, such as when tools are used to gauge the accuracy with which the device reproduces an original sound; or it can be measured subjectively, such as when human listeners respond to the sound or gauge its perceived similarity to another sound.

<span class="mw-page-title-main">Meridian Lossless Packing</span> Audio file format

Meridian Lossless Packing, also known as Packed PCM (PPCM), is a lossless compression technique for PCM audio data developed by Meridian Audio, Ltd. MLP is the standard lossless compression method for DVD-Audio content and typically provides about 1.5:1 compression on most music material. All DVD-Audio players are equipped with MLP decoding, while its use on the discs themselves is at their producers' discretion.

Full Rate was the first digital speech coding standard used in the GSM digital mobile phone system. It uses linear predictive coding (LPC). The bit rate of the codec is 13 kbit/s, or 1.625 bits/audio sample. The quality of the coded speech is quite poor by modern standards, but at the time of development it was a good compromise between computational complexity and quality, requiring only on the order of a million additions and multiplications per second. The codec is still widely used in networks around the world. Gradually FR will be replaced by Enhanced Full Rate (EFR) and Adaptive Multi-Rate (AMR) standards, which provide much higher speech quality with lower bit rate.

<span class="mw-page-title-main">DTS (company)</span> Series of multichannel audio technologies

DTS, Inc. is an American company. DTS company makes multichannel audio technologies for film and video. Based in Calabasas, California, the company introduced its DTS technology in 1993 as a competitor to Dolby Laboratories, incorporating DTS in the film Jurassic Park (1993). The DTS product is used in surround sound formats for both commercial/theatrical and consumer-grade applications. It was known as The Digital Experience until 1995. DTS licenses its technologies to consumer electronics manufacturers.

<span class="mw-page-title-main">DTS-HD Master Audio</span> Lossless audio codec for home theater

DTS-HD Master Audio is a multi-channel, lossless audio codec developed by DTS as an extension of the lossy DTS Coherent Acoustics codec. Rather than being an entirely new coding mechanism, DTS-HD MA encodes an audio master in lossy DTS first, then stores a concurrent stream of supplementary data representing whatever the DTS encoder discarded. This gives DTS-HD MA a lossy "core" able to be played back by devices that cannot decode the more complex lossless audio. DTS-HD MA's primary application is audio storage and playback for Blu-ray Disc media; it competes in this respect with Dolby TrueHD, another lossless surround format.

SBC, or low-complexity subband codec, is an audio subband codec specified by the Bluetooth Special Interest Group (SIG) for the Advanced Audio Distribution Profile (A2DP). SBC is a digital audio encoder and decoder used to transfer data to Bluetooth audio output devices like headphones or loudspeakers. It can also be used on the Internet. It was designed with Bluetooth bandwidth limitations and processing power in mind to obtain a reasonably good audio quality at medium bit rates with low computational complexity. As of A2DP version 1.3, the Low Complexity Subband Coding remains the default codec and its implementation is mandatory for devices supporting that profile, but vendors are free to add their own codecs to match their needs.

<span class="mw-page-title-main">Audio coding format</span> Digitally coded format for audio signals

An audio coding format is a content representation format for storage or transmission of digital audio. Examples of audio coding formats include MP3, AAC, Vorbis, FLAC, and Opus. A specific software or hardware implementation capable of audio compression and decompression to/from a specific audio coding format is called an audio codec; an example of an audio codec is LAME, which is one of several different codecs which implements encoding and decoding audio in the MP3 audio coding format in software.

<span class="mw-page-title-main">LDAC (codec)</span> Digital audio encoding technology

LDAC is a proprietary audio coding technology developed by Sony, which allows streaming high-resolution audio over Bluetooth connections at up to 990 kbps at 32 bits/96 kHz. It is used by various products, including headphones, earphones, smartphones, portable media players, active speakers, and home theaters.

Low Latency High-Definition Audio Codec (LHDC) is an audio codec technology developed by Savitech. LHDC allows high-resolution audio streaming over Bluetooth. It is a high-quality Bluetooth codec based on the A2DP Bluetooth protocol and allows a bit-rate of up to 900 kbit/s compared to SBC's bit rate of 345 kbit/s.

L2HC is a proprietary audio codec technology developed by Huawei. L2HC allows high-resolution audio streaming over Bluetooth and NearLink with lossless data compression of digital music. It is a high-quality Bluetooth based on the A2DP Bluetooth protocol and NearLink based on its native L2HC 3.0+ StarLink protocol with Polar protocol by Huawei and allows a bit-rate of up to 1.5Mbps compared to SBC's bit rate of 345kbps and LDAC's bitrate of 960kbps.

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