While audiophiles and poor people alike decried the death of the beloved headphone jack in smartphones, this was a source of an unexpected windfall for the Bluetooth Special Interest Group (SIG). Apple’s vulgar pursuit of minimalism in smartphone design had effectively replaced the cheap and reliable copper wire between headphones and smartphones with expensive and finicky Bluetooth wireless hardware.
Meanwhile, the Bluetooth SIG laughed all the way to the bank with the $25,000 in licensing and certification fees it collects per product, as an increasing number of smartphone manufacturers resorted to the familiar monkey see, monkey do routine of aping Apple’s “brave” new design choices.
Whether you like it or not, Bluetooth is the unsavoury present and the potentially even less savoury future of portable audio. Familiarising yourself with the complicated workings of Bluetooth audio is essential to navigating the jargon-riddled mess of purchasing a pair of wireless earphones these days. Fortunately, we will be distilling Bluetooth audio down to its basics to help you make informed purchase decisions.
Bluetooth Wireless Audio Is Complicated
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Nothing beats the simplicity and efficiency of hooking up earphones to smartphones with wires. The free flow of electrons through the copper strands doesn’t limit the sound quality, range, and power consumption of devices attached to either end of the wire. In addition to being cheap as dirt and uncomplicated, wired audio connections are free from interference and latency issues plaguing their wireless counterparts.
Transmitting audio wirelessly, however, is a deceptively expensive and complicated affair. Doing so requires the conversion of audio signals into radio waves and back, which entails expensive radio hardware that’s usually one of the most expensive components in typical wireless earphones. The expense is doubled because the radio hardware is required on both the source (smartphone) and sink (earphone) ends of the wireless audio equation.
You’re sorely mistaken if you consider the speaker as the most expensive component of an earphone because that spot is reserved for the DSP (Digital Signal Processing) hardware required to make the audio signal light enough to be pumped into the airwaves.
Such expensive hardware is needed because the most reliable medium of consumer wireless transmission, the 2.4GHz radio spectrum, is populated with competing signals emanating from devices ranging from Wi-Fi routers and smart home devices to baby monitors and microwave ovens. Transmitting audio wirelessly in its raw, uncompressed form would not only clog the radio waves, but the high bandwidth requirement would also make it more vulnerable to interference and signal loss.
Every wireless audio protocol, such as Bluetooth audio, must therefore compress the audio signal to the maximum possible extent before transmitting it through the crowded 2.4GHz spectrum. Higher data compression levels require more powerful processing hardware, while also adding a significant amount of latency to the audio signal.
This is precisely why watching movies with the audio routed wirelessly over Bluetooth headsets or speakers often leads to lip-sync issues.
In a nutshell, it is difficult to reliably transmit high bandwidth audio over the 2.4GHz radio spectrum, but audio compression can mitigate this issue. However, the added processing required to compress and decompress audio signals is not only expensive but also causes latency and lip-sync problems. Knowing how these mutually exclusive attributes work is essential to understanding the pros and cons of various Bluetooth audio formats.
How Is Music Turned Into Digital Audio?
At the simplest level, the human sense of hearing boils down to the eardrums sensing the pressure differences created by sound waves propagating through air. In other words, music played from the loudspeaker is essentially a specific pattern of air pressure differentials that we interpret as a familiar musical composition.
We hear high-frequency notes when the speaker cone vibrates at a high rate, whereas the same vibrating at a lower rate is perceived as bass (low frequency) notes.
This information is relayed to speakers or earphones as a series of sinusoidal waveforms. These waveforms, in turn, represent the pattern of electrical impulses required to drive the speaker cone at just the right combination of frequencies and amplitudes to create desired sound notes. However, modern devices such as personal computers and smartphones cannot process sound in the analogue domain.
These analogue waveforms must be transformed into digital signals by an analogue-to-digital converter (ADC) before they can be processed by your PC or mobile device. Once processed, these digital signals are then converted back to analogue waveforms by a digital-to-analogue converter (DAC).
But because computers aren’t competent at drawing curves, these sinusoidal waveforms are chopped up into a series of segments. Each segment is then encoded by the ADC into digital data that can be processed by the music player before being reconstructed back into analogue waveforms by the DAC and fed into the speakers or earphones as electrical impulses.
The waveform in each segment is represented by data points mapped in terms of time and amplitude across the X and Y axes, respectively. The gaps in information between these data points are approximated to recreate the waveform as closely to the original as possible. Greater the number of data points, the better the accuracy of the recreated waveform. This, in turn, translates into better audio quality.
Understanding Bluetooth Audio Jargon
Knowing how audio information is represented in the digital domain is key to making sense of Bluetooth audio jargon. The practical difference between Bluetooth audio formats is down to the degree of compression achieved, whether the compression algorithms discard data (lossy) or not (lossless), and finally the maximum bandwidth of audio data transmitted wirelessly.
Let’s take a closer look at key Bluetooth audio format specifications such as bit depth, bit rate, and sample rate, while learning how these affect audio quality and ease of wireless transmission.
What Is Sample Rate?
The sample or sampling rate of an audio format is measured in hertz (Hz) and represents the number of data points present per second of the audio signal. Since these recorded data points are used to recreate the original analogue waveform passed onto the speakers, a digital audio file encoded at a high sample rate sounds closer to the original audio recording when played back through your music system.
Is High Sampling Rate Worth It?
A higher sampling rate significantly increases data density. The consequent increase in filesize makes it difficult to stream the audio recording over a Bluetooth connection. Thankfully, there’s a point beyond which humans can’t perceive the improvement in quality gleaned from increasing the sample rate.
This limit is a consequence of a digital audio encoding method called pulse-code modulation (PCM), which requires the sample rate to be approximately twice that of the highest audio frequency meant to be encoded faithfully. Since the human hearing is capped at 20,000Hz (or 20kHz), the improvement in audio quality drops off significantly after a sample rate of 40kHz.
Not surprisingly, older analogue professional audio equipment used sampling rates in the range of 40kHz to 50kHz. That’s also why CD and DVD audio is sampled at 44.1kHz and 48kHz, respectively.
However, modern audio formats use much higher sample rates of 96kHz and even 192kHz. Some suggest that this leads to a higher audio fidelity, whereas others argue that encoding ultrasonic frequencies inaudible to humans can cause intermodulation distortion upon playback.
If that wasn’t confusing enough, the delta-sigma modulation approach to digital audio encoding employs pulse-density modulation (PDM) instead of PCM. This involves sampling rates in the order of megahertz, but at a resolution of a mere 1-bit. For example, the 1-bit delta-sigma encoding found in the SACD format can reproduce a frequency response of 100kHz.
This is a good illustration of how specifications can vary wildly based on the encoding method used in the audio format.
What Is Bit Depth?
If the 1-bit part got you intrigued back there, it represents another important specification relevant to digital audio formats. Bit depth denotes the individual resolution of each data point captured in the sampling process and is measured in terms of the number of digital bits employed to represent each audio sample.
A higher bit depth significantly increases the accuracy of the recorded audio signal, because that allows a greater amount of data to be recorded per sample. In fact, encoding accuracy doubles for every 1-bit increase in bit depth. Without going too much into the weeds of audio theory, the bit depth of an audio format has a direct impact on the signal noise and the dynamic range of the recording.
How Many Bits Do You Need for Great Sound Quality?
Short answer: 16-bit.
Long answer: Higher bit depth reduces signal noise by incorporating more accurate information of the audio waveform, which reduces the guesswork needed to recreate it. This consequently decreases quantisation noise, or the errors created by digital approximation.
The digital recreation of an analogue waveform requires approximation because it is impossible to use finite coordinates to represent sinusoidal curves made up of infinite points.
This has a direct impact on the signal-to-noise ratio (SNR), which is measured in decibels (dB). You ideally want the SNR to be higher than the total dynamic range of the source audio, or you will hear distortion during audio playback. Most modern music genres exhibit a dynamic range of 60dB, whereas Western Classical recordings need a much higher range at 70dB.
That’s why an 8-bit audio recording, with an SNR of 48dB, sounds perceptibly lower quality compared to a 16-bit one, which clocks in at 96dB.
It must be noted that SNR of an audio format is also roughly equal to its dynamic range. Most modern audio formats are encoded at 24-bit, which amount to a dynamic range of 144dB. However, the theoretical limit of human sensitivity to sounds peaks at 120dB, which is roughly equivalent to 20 bits.
In reality, the practical capability of the human ear to perceive dynamic range is much lower, so a bit depth of 16 bits is more than enough to accurately reproduce music of all genres.
What Is Bit Rate? What Separates It From Bit Depth?
The maximum possible bit rate of an audio format is a measure of the amount of data that can be transferred per second. It is the multiplicative product of the sample rate and bit depth expressed in kilobytes or megabytes per second (kbps/Mbps). A higher quality audio recording encoded with a large sample rate and high bit rate will amount to a significantly higher bit rate.
What Are Audio Codecs?
Most digital audio formats used in legacy physical media such as CD, SACD, DVD, and Blu-ray are of the uncompressed kind. Such audio formats reap the benefits of transmitting audio without any quality loss, processing overhead, or latency associated with audio compression. However, this comes at the cost of excessive space requirements.
While this may not be a concern for physical media, you don’t want to transfer large amounts of data wirelessly over Bluetooth if you can help it. This is where audio compression comes into play. In the audio realm, this is achieved by utilising codecs such as MP3, AAC, Ogg Vorbis, WMA, and FLAC.
The word codec is a portmanteau of coder/decoder, which also represents its use case of encoding and decoding digital audio.
|Uncompressed Audio||Lossless Compression||Lossy Compression|
|Pros||• No data lost to compression|
• Universal compatibility
|• Retain audio data|
• Reduced processing time
|• Smallest filesize|
• Ideal for wireless transmission
|Cons||• Requires high data bandwidth|
• Not feasible for wireless transmission
|• Poor compression ratio||• Doesn’t retain the entire original audio signal|
|Audio Formats||WAV, AIFF, LPCM||FLAC, ALAC||MP3, AAC, WMA|
How Does Audio Compression Work?
Audio codecs use a combination of sophisticated algorithms and advanced mathematics to find clever ways of shrinking down uncompressed data. This usually involves leveraging mathematical formulas to identify data patterns and using the magic of advanced mathematical models to represent the same in a smaller data storage footprint.
Codecs that shrink down data without discarding any information are called lossless codecs. Such compression algorithms can recreate the original audio without perceivable quality loss. FLAC is one of the most recognisable examples. Although lossless compression is great for sound quality, it doesn’t do wonders for keeping the file size and bit rate low enough for Bluetooth wireless transmission.
This is where lossy codecs such as MP3 and AAC make a case for themselves.
In addition to employing clever algorithms and advanced mathematics to compress data, these codecs also leverage the unique way in which humans perceive audio. Due to the idiosyncrasies of the human ear and the way our brains processes sound, our perception of the audio frequency spectrum isn’t uniform.
In fact, an entire branch of science called psychoacoustics exists to study this phenomenon. Lossy codecs employ the principles of psychoacoustics to discard audio information that isn’t perceptible to most humans. Such formats achieve a significant degree of compression by eliminating frequencies that are otherwise inaudible.
The accuracy of such psychoacoustic algorithms is surprisingly good, with most listeners failing to tell the difference between music encoded in lossless codecs and lossy codecs such as MP3/AAC.
This Is Why Bluetooth Needs Real-Time Codecs
Both lossy and lossless codecs such as WMA, MP3, Ogg Vorbis, and FLAC originate from an era when music was either downloaded or transferred over physical mediums such as flash drives and compact discs. These lossy codecs had no reason to hold back on complex compression algorithms despite their high processing overheads and encoding/decoding time.
Unfortunately, these are luxuries of the bygone wired era that wireless Bluetooth audio codecs cannot afford.
For starters, Bluetooth audio involves on-the-fly compression of audio from the source device before it is streamed to a pair of True Wireless Stereo (TWS) earphones or wireless speakers. Employing computationally expensive compression algorithms increases the overall processing latency, which causes lip-sync issues. The high processing overheads associated with these compression algorithms also has a negative impact on the battery life of wireless devices.
Bluetooth employs codecs optimised for real-time streaming needs such as SBC, aptX, and LDAC to strike a balance between compression efficiency and encoding/decoding overheads. These codecs achieve this feat by resorting to simpler noise shaping compression algorithms instead of using computationally expensive and slower psychoacoustic alternatives.
Noise shaping algorithms might not be as space-efficient, but real-time Bluetooth codecs get around this problem by either compromising on audio quality or by increasing the bit rate of the audio stream. Not surprisingly, Bluetooth codecs cannot match the sound quality of their offline counterparts for similar bit rates.
Various Bluetooth Audio Codecs Compared: SBC, AAC, aptX, aptX HD, aptX LL, aptX Lossless, SSC, LDAC, LHDC, LC3
Now that we know how wireless audio works better than most audio journalists, let’s take a look at the various popular Bluetooth audio codecs and what compromises they make to compression, latency, bit rate, and audio quality to stream audio effectively over thin air.
|Codec||Max Bit Rate||Max Bit Depth||Max Sample Rate||Variable Bit Rate|
|aptX HD||576kbps||24 bit||48kHz||No|
|aptX Adaptive||280-420kbps||24 bit||48kHz||Yes|
|aptX LL||352kbps||16 bit||44.1kHz||No|
|aptX Lossless||1Mbps||24 bit||96kHz||No|
|LDAC||990kbps||24 bit||96kHz||Yes (Manual)|
|Samsung Scalable Codec||512kbps||24 bit||96kHz||Yes|
1. SBC (Low-Complexity Sub-Band Codec)
SBC or low-complexity sub-band codec, as its name suggests, is the baseline codec available for free to all Bluetooth SIG members. Like most things free, it’s designed to maximise compatibility over audio quality or wireless streaming performance. It is generally found in budget Bluetooth devices, especially where the hardware vendors lack the budget to license more efficient third-party codecs.
It might not possess the most efficient compression algorithm, but it can still deliver audio sampled at 48kHz at a bit rate of 345kbps over Bluetooth. However, for the same bit rate, SBC will deliver lower audio fidelity compared to virtually every other real-time Bluetooth codec. The higher bandwidth required to achieve an acceptable level of audio quality also means that SBC audio streams are prone to connection issues.
This is also the codec your Bluetooth connection will default to when either the source or the sink Bluetooth device doesn’t support other third-party audio codecs.
2. aptX, aptX HD, aptX LL, aptX Adaptive, and aptX Lossless
During the course of this guide, we have learned the mutually exclusive relationship between audio quality, compression efficiency, bit rate, and signal latency. It’s hard to find the right balance between these parameters. Qualcomm’s aptX family of Bluetooth audio codecs, however, gets around this conundrum by offering a smorgasbord of options optimised for your specific Bluetooth audio use case.
Need the quickest possible audio feedback while playing games or watching movies? You might want to try out the aptX LL or Low Latency codec that makes some compromises in the audio compression department to deliver latency of 40 milliseconds or quicker.
Music connoisseurs, who aren’t concerned with lip-synchronisation, are served better by aptX HD that supports high fidelity 24-bit/48kHz LPCM audio at 576kbps. That’s equivalent to DVD quality audio.
However, true audiophiles can rejoice with the relatively recent announcement of the aptX Lossless codec. It lets you choose between lossy 24-bit/96kHz audio stream and true lossless, bit-for-bit accurate audio encoded at 16-bit/44.1 CD quality. Either way, this codec can achieve a bit rate of a whopping 1Mbps.
Even the plain vanilla aptX codec is capable of delivering 16-bit/48kHz fidelity up to a bit rate of 352kbps, which is significantly better than SBC. Qualcomm’s codec is also known for keeping latency in check by using a less processor-intensive compression algorithm that works by encoding the sequential difference between audio samples within the stream.
If you can’t decide between these aptX flavours, you might want to take a gander at aptX Adaptive. Qualcomm seems to be intent on replacing plain vanilla aptX with this smarter version, but it hasn’t trickled down in too many devices since its 2020 release. It can automatically tweak compression efficiency, bit rate, and audio encoding quality to suit the appropriate use cases by adjusting bit rates between 280kbps and 420kbps, while also keeping the latency contained between 50ms and 80ms.
Products supporting aptX must meet Qualcomm’s minimum quality requirements, which makes the experience reliable while ensuring inter-compatibility. This also makes aptX unavailable to iPhone users for obvious reasons.
3. AAC (Advanced Audio Coding)
Speaking of Apple, AAC is quite similar to the age-old MP3 codec, albeit being slightly better in terms of audio quality and compression. It is also one of few Bluetooth audio codecs (apart from the rare few Bluetooth devices supporting MP3) that works over Bluetooth despite the fact that it isn’t optimised for real-time encoding.
That also means it suffers from higher latency and causes greater battery drain due to the steep computational requirements of its advanced psychoacoustics-based compression algorithm. Not surprisingly, AAC can achieve decent audio quality at a bit rate of just 250kbps. However, the lack of a certification process also means the implementation is spotty on Android devices.
4. LC3 (Low-Complexity Communications Codec)
Unveiled at the CES 2020, the new Bluetooth Low Energy audio standard embraces a brand-new codec dubbed LC3 (Low-Complexity Communications Codec). It was co-developed in conjunction with Fraunhofer IIS, with the goal of improving upon the SBC codec by delivering higher audio quality while also keeping battery consumption low.
The new compression algorithm devised by the researchers at Fraunhofer IIS claims to cut down SBC codec’s maximum bit rate of 345kbps down to a mere 192kbps, while also improving on sound quality and reducing the power requirements at the same time. Qualcomm claims this allows the Bluetooth LE Audio standard to improve the range and battery life of compatible Bluetooth devices and accessories.
However, the capability to have multiple audio streams is one of the coolest improvements that comes with Bluetooth LE Audio and LC3 codec. We have discussed this in detail in our comprehensive Bluetooth guide, but it basically allows one-to-many and many-to-one audio streams. In other words, multiple Bluetooth audio devices can play audio from a single source, while also allowing multiple sources to simultaneously serve several audio streams to a single receiver.
Other cool improvements include the capability to wirelessly deliver stereo channels as two separate audio streams. Traditional Bluetooth audio involves a single audio stream transmitted to a pair of TWS earphones, which requires another Bluetooth link between the left and right earphones to enable discrete stereo channels.
A lot has been promised with Bluetooth LE Audio, and the list of features definitely looks impressive, but the standard hasn’t made its way into mainstream mobile hardware as of this writing.
The only thing harder than figuring out what the acronym LDAC stands for is trying to find compatible Bluetooth devices supporting Sony’s proprietary codec. The company’s own smartphones and audio devices come with LDAC support, as do popular enthusiast-grade PMPs (portable media players), and select third-party smartphones. But the selection of devices that supports this codec is meagre despite LDAC being a part of the AOSP (Android Open Source Project) for quite some time.
LDAC differs from most other codecs with its variable bit rate feature. Its baseline 330kbps is good for ensuring connection reliability, but this comes at the cost of sound quality. In fact, both SBC and aptX outperform LDAC at this preset bit rate. This is the default quality preset, sadly, which can only be changed by rummaging through the phone’s settings menu. For all practical purposes, most laymen will experience the worst possible experience with LDAC unless they make a concerted effort to change the default bit rate.
The two improved sound quality modes deliver 660kbps and 990kbps, with the latter capable of pumping out audio at 16-bit/48kHz. These modes can keep up with Qualcomm’s aptX HD, which explains why so many PMPs lacking Qualcomm hardware have support for LDAC due to its AOSP roots.
6. LHDC (Low-Latency and High-Definition Audio Codec)
Developed in partnership by the Hi-Res Wireless Audio (HWA) Union and Taiwanese audio hardware solutions company Savitech, the LHDC codec’s acronym stands for low-latency and high-definition audio codec. As its name suggests, it triples the audio data transmission capability of SBC by pumping out audio signals at 900kbps.
Unlike LDAC and most other codecs, LHDC can support sample rates up to 96kHz. But like LDAC, LHDC is also a part of the AOSP and therefore supported by Android 10 devices and newer. Although the Huawei Mate 10 was the first smartphone to support the format, the uptake has been slow in the mobile audio hardware space.
The audio quality-focused LHDC codec is complemented by the low-latency audio codec (LLAC), which sacrifices some of the audio fidelity to achieve lower latency necessitated by gaming and audio playback associated with motion picture. The codec claims to achieve a minimum latency of 30ms, while reducing the bit rate down to a range of 400kbps to 600kbps at the audio resolution of 24-bit/48kHz. Huawei P30 was the first phone to support LLAC.
7. Samsung Scalable Codec
While LDAC offers the choice between three different bit rates, which is great for lowering audio bandwidth in order to maintain audio signal integrity in environments with greater radio noise, you have to do that manually. The Bluetooth SIG’s new LC3 codec and Qualcomm’s existing aptX Adaptive codec can adjust bit rate automatically on the fly.
However, Qualcomm isn’t the only one with this capability, because Samsung had developed its proprietary Scalable Codec in conjunction with audio expert AKG for the express purpose of enabling variable bit rate streaming for the Galaxy Buds. This codec analyses the airwaves for Wi-Fi interference and automatically adjust the bit rate from 88kbps to 512kbps to maintain consistent Bluetooth audio playback.
Codecs Matter, But Hardware Matters More
We have dissected in great detail how the various specifications of Bluetooth audio codecs affect audio quality, latency, and signal reliability. You are now well equipped to choose the best Bluetooth codec to suit your particular needs. But at the end of the day, the choice squarely depends on your hardware ecosystem.
Unlike Android users who are spoiled for choice when it comes to Bluetooth codecs, iPhone users must make do with the whims of Apple and its love for AAC. And even though Android users can choose a codec of their liking, that only takes care of the audio link between their Bluetooth audio accessories and the phone/PMP. They still have little control over the codec used by the music and movie streaming sites.
But thankfully, as long as you stay away from SBC, the practical difference in perceptible audio quality between modern Bluetooth codecs is minimal. What matters more is the enclosure material, design, and electronics hardware used in your Bluetooth audio equipment
Your first priority should ideally involve selecting the best Bluetooth audio hardware, with codec choice coming in later. Because the former has a far greater impact on the way your music sounds.