Our perspective of the Best Sound

Our philosophy is that a great sound system should faithfully reproduce the original sonic characteristics as perceived by the human auditory system. This may seem simple, but it is much more challenging than it appears.

Imagine a perfect microphone with a flat frequency response from 20Hz to 20kHz (the human hearing range) that can capture every sound with accurate dynamic levels. Paired with a perfect speaker that has an identical flat response, the inputs would be identical to the outputs - what you hear from the source is exactly what you get from the speaker. This represents the definition of the best possible sound quality.

Numerous reputable studies have shown that a flat frequency response is a key criterion for delivering the most enjoyable sonic experience to listeners. Loudspeaker design and manufacturing prioritize achieving this flat response. However, when we measure the system in a real-world listening room, it is nearly impossible to obtain a truly flat response.

The published frequency response specifications of speakers are typically measured on the "acoustic-center axis" (in-phase plane) in an anechoic chamber (without reflections). This is done to eliminate the effects of the surrounding environment and off-axis signal interference. But once the sound interacts with reflective surfaces in a real room, the measurement becomes "contaminated." Since we cannot control the user's listening environment, we cannot standardize a single target response - each room has unique acoustic characteristics.

These environmental factors significantly alter the direct sound from the speaker before it reaches the listener's ears. There are simply too many variables to track and account for, making it almost impossible to maintain the original sound fidelity.

No matter how much you invest in the sound system or room acoustics, the sound quality will be compromised to some degree, preventing the system from reaching its full potential. It's like having too many bright lights spotlighting a picture, and we need to reverse-engineer the painting to retrieve the true, original colors.

We understand that objectively measuring sound quality is challenging, as auditory perception is highly subjective. People have lost faith in the calibration process because, when not done correctly, it can make the sound quality worse than leaving the system uncalibrated.

After over 20 years of studying the science behind these sound alterations, we have developed the knowledge and specialized techniques to align measurement data with human perception. We call this the "Timbre Chart Tuning” method, and we are excited to share this outstanding sound quality solution with the world.

Our comprehensive approach to achieving the best sound quality focuses on three main factors:

Propagation behavior differences of sound sources
Reflections mixed with direct sound
Spectral/spatial shifts due to speaker directivity and throw distance

To address these factors, we employ four key strategies:

Spectral balance
Phase consistency
Sonic (stereo) image reconstruction
Coverage optimization

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We will explain the science behind these in our Timbre Theory.

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TIMBRE THEORY (COLOR THEORY IN SOUND)

Timbre Theory (Color Theory in sound)

One day, I watched my 3-year-old daughter playing with clay, mixing and rolling different colors together into a small planet. When I asked her what her favorite color was, she would always say "rainbow." I think this is because rainbows display almost all the colors we can see in the visible spectrum, giving us the fullest color contrast and the most beautiful phenomenon that most people can agree on.

The same principle can be applied to audio. To reproduce the highest sound quality for listeners, a good sound system must not only be able to reproduce all the frequencies we can hear across the audible spectrum, but it must also deliver those frequencies accurately in terms of loudness level and phase relationship. The question is, how do we determine what "accurate" means for human perception?

Off-sweet-spot response

It's not uncommon to find speakers in the market that have an excellent frequency response specification. However, people often complain that these "good response" speakers don't actually deliver good sound quality in real-world listening scenarios.

The reason for this is that the ideal, flat frequency response is typically only achieved within a very narrow "sweet spot" - a small, specific listening position. The published frequency response measurements are usually taken at 1-2 meters on the speaker's acoustic axis, in an anechoic chamber. But manufacturers cannot customize each speaker to fit the unique listening angles and distances in every user's environment.

That's why speakers require calibration to perform well in different environments; they need specialized tuning to deliver their full potential outside of the controlled anechoic conditions. The flat frequency response specification alone does not guarantee good sound quality when the speaker is placed in a real-world listening room with reflections and other acoustic variables.

Why the Off-sweet-spot response is important too?

In a real-world listening environment, we don't just hear the direct sound coming straight from the speakers. We also hear the reflected sound bouncing off the surfaces in the room. The direct sound can be manipulated, but the reflected sound is largely out of our control.

The direct and reflected sounds become mixed and altered together before reaching our eardrums. The speakers act like the strings of a guitar, while the room functions as the guitar's resonant body, becoming an integral part of the overall sound.

What's interesting is that our brains can differentiate between the direct and reflected sounds to some degree, but not fully. It's hard to determine exactly how much of each component is mixed in our perception, and to what extent our hearing system can separate them.

The more important question becomes: Can we control the final combined sound by manipulating the input signals before they mix in the room? In other words, can we pre-compensate for the acoustic interactions to achieve the desired overall sound?

What makes the Off-sweet-spot response different at the first place?

The nature of the sound source itself is a key factor that causes sound propagation to behave differently at various angles, distances, and after reflections. The physical size of the sound source means that the frequencies in the spectrum have their own distinct directivity patterns and intensity attenuation rates.

Furthermore, the arrangement or positioning of multiple drivers within a loudspeaker, as well as the geometrically unstable reflections in the room, make these propagation behaviors even more complicated and unpredictable, especially in the crossover frequency ranges where the drivers are transitioning.

It is simply impossible for us to track and account for all of these complex interactions happening in a real-world listening environment. The factors influencing sound propagation are too numerous and difficult to fully predict or control.

RGB Spot Lights Analogy (See pic)

Let's use a visual analogy to illustrate the audio challenges we're addressing. Imagine there are three spotlights of different colors (red, green, and blue) shining on a projection screen in a room. These spotlights represent the individual drivers (woofer, midrange, tweeter) in a speaker system, and the screen represents the listening area.

Each light has its own inherent beam angle, just as audio frequencies have different directivity patterns. To emit a perfect white color at the central "sweet spot" on the screen, the intensities of these three lights must be perfectly balanced. This is akin to achieving the ideal spectral balance in the on-axis listening position.

However, if we look at the outer edges of the spot (off-axis), we start to see color distortion, fading into yellow, and ultimately turning red. This demonstrates why it's so critical to listen at the acoustic center of the speaker, where the perfect spectral balance occurs.

Now, let's make the situation worse. Imagine the room interior is no longer black, but instead painted in various random colors, mostly in red (representing no low-frequency absorption). These colored surfaces cause the light rays to reflect back to the screen (the listening area).

As a result, we no longer get the perfect white color at the central aiming point on the screen, and the colors shift elsewhere. In audio terms, this is where we need to use equalizers and filters to rebalance the frequency responses and retrieve the desired, accurate sound.

Reflections

In the visual analogy, it's relatively easy to paint the room black to absorb all the reflected light. Unfortunately, in the audio realm, it is much more challenging to absorb all the sound energy, especially the low frequencies, even in professional studio environments.

The low energy and narrow directivity of high frequencies make them less problematic to manage. However, the high power and omnidirectional nature of low frequencies make them extremely difficult to control. These low frequencies will continue bouncing around the room without much ability to contain them.

Linear Phase

Many people claim that we cannot perceive phase differences in sound. This statement is only true when the phase is already non-linear to begin with. Unless the speaker drivers are coaxial, the sounds from multiple drivers can only be in-phase at a very thin, specific listening plane - the acoustic center.

Off this central plane, the travel time of sounds from the different drivers to the listener's ears will be different. Factoring in the drivers' directivity patterns and room reflections, you end up with a directivity mismatch, which causes issues in the frequency response both on-axis and off-axis. At this point, the phase response is no longer linear and has become randomized. This results in a sound that has an electrically "reverberated" quality.

That's why when you play back a voice recording through a typical speaker system, it often doesn't sound like a real person speaking in front of you. The spatial cues and phase coherence are lost. The "auditory proximity" is absent.

Auditory Proximity

It's difficult to convey to people what good sound quality is without giving them the opportunity for direct A/B comparisons. The concept of "auditory proximity" is not widely discussed in the audio industry, but we believe it is closely tied to sound quality.

As coaxial or linear-phase speaker designs become more popular, people may start to appreciate the importance of this feature, especially as the current trend towards immersive audio requires such phase coherence. In our opinion, achieving this linear phase response, like that of the original raw sound source, is crucial for recreating a sense of auditory proximity.

The human ear detects this auditory proximity through the phase coherence of upper harmonics in the direct sound, which are randomized by early reflections. Preserving the harmonic phase is an essential element of reproducing realistic, natural-sounding audio.

Timbre Chart TUNING METHOD

Color Chart in Audio

If the primary sound quality issue with an audio reproduction system is the inconsistent frequency response across different listening environments with varying reverberation patterns, then maintaining a flat frequency response (FR) across all components in the signal chain would be the right approach. However, this FR maintenance must be relative to our human hearing, not just the analysis equipment, as the measurement tools do not perfectly align with our auditory perception.

Using our auditory system to flatten the responses is the key!

The first thing we need to identify is our individual frontal hearing response; it is definitely not flat, and it varies to some degree between people, much like fingerprints. These frontal and non-frontal responses also change depending on the sound direction.

Fortunately, we do not need to tune the system to match our individual ears; we are simply aiming to flatten the sound as it would naturally be in an anechoic (non-reflective) environment. We are utilizing our hearing mechanism to determine what the anechoic flat responses should sound like. We just need to focus the frequency domain analysis on our auditory system.

This approach is similar to color calibration for display screens. There are clever techniques that simplify the process for users without requiring deep color management knowledge. The idea is to provide a set of graphic color/brightness charts and simple instructions for users to create an accurate profile for their own displays.

Our Timbre Chart Tuning method takes a similar approach in audio. We have created a set of test tones designated for our well-trained engineers' frontal hearing. It acts like a personalized "color chart" in audio, allowing us to identify the true relative levels of each frequency band. Combined with other deep measurement tools, like specific time-windowed dual-FFT techniques, we can then recreate the best frequency response that the system is designed to deliver.

Target Curve Vs Flat Curve

Suppose your ultimate goal is to achieve the best sound quality that faithfully reproduces the original nature of the sound before it was captured, in terms of human hearing perception. The flat frequency response curve is what you are aiming for, but this can only be achieved in anechoic conditions, where you are solely hearing the direct sound.

Hypothetically, if we could maintain a flat direct sound response in every sound system, from recording to mixing to consumer playback, and listen to them in anechoic environments, the inputs would be exactly equal to the outputs, and there would be no "circle of confusion" problem.

However, in a typical reverberant listening room, when we analyze the ideal direct sound using steady-state measurements, we notice some significant issues. The bass frequencies are boosted by more than 18dB, depending on the room size and speaker placement, as their long wavelengths and fast propagation cause constructive interference. The direct bass has accumulated low-frequency reflections.

Meanwhile, the high frequencies may exhibit an attenuation rolloff, mainly due to the comb filtering effect (acting like a negative feedback loop) and the narrow directivity of high-frequency drivers. Perceptually, we need to attenuate the bass and boost the treble, but the challenge is that we don't know how much to adjust, as the reflections are already mixed with the direct sound, and we don't know the human hearing time window for differentiating them.

Counterintuitively, flattening the steady-state frequency response will weaken the bass and overshoot the treble, leading to a piercingly bright sound. This is why people often prefer to do nothing rather than targeting the flat response curve. Some take an alternative approach, using different target curves to predict how much low and high frequencies need to be boosted or suppressed to match the perceived response of the accumulated direct and reflected sound.

Unfortunately, due to the unique reflection patterns in every room, there is no "one target curve fits all" solution.

Fast-Time Windowing (FTW)

What about using the Finite Time Window (FTW) measurement technique? Can this help solve the problems we've discussed?

The FTW approach is to limit the measurement time to a specific, short duration. The goal is to avoid capturing the reflections by removing the later-arriving sound energy. This brings the measurements closer to representing just the direct sound.

However, even with FTW, we still cannot separate the reflections in the same way our ears and auditory system can. Additionally, it is controversial to claim that FTW can effectively reduce the detrimental artifacts caused by reflections in the frequency response, such as comb filtering.

That said, utilizing the FTW is still a valuable reference for spectral balancing, especially in the low-frequency range. By focusing the measurement on just the direct sound, it provides helpful insights for tuning the system.

What’s Sonic (Stereo) Image?

The purpose of designing a stereo/multi-channel sound system is to create a Sonic (Stereo) Image

(except the LFE or other effect channels).

Just as light entering our eyes from different angles creates a visual image in our brain, sounds entering our ears from various directions create a sonic image in our auditory perception.

In 1931, engineer Alan Blumlein at EMI had an idea to address the issue he observed at a movie theater, where the actor's voice seemed to come from a different location than the visual image on the screen. Blumlein invented multi-channel stereophonic technology, including stereo records, stereo films, and surround sound.

To this day, all the advanced sound systems in the commercial market, including immersive audio technologies, are still fundamentally utilizing Blumlein's principles. The key differences that impact the final sound quality lie in the rendering solutions, speaker configurations, and calibration processes, which should not be underestimated.

In our RGB spotlight analogy, a single-channel signal would provide a single spot on the screen, while multiple channels would create multiple spots. These spots act like the pixels of a soundscape, constructing what we call the Sonic Image; more channels allow for a higher resolution of this image. Unlike visual perception, our binaural hearing system doesn't require very high resolution to form a Sonic Image, as we can utilize the simple principle of "Amplitude Panning" (also known as "Amplitude Panoramic Perception").

Even a two-channel stereo configuration can provide a good sensation of soundscape and envelopment. However, as the trend of immersive audio evolves, the design of multi-channel speaker configurations has become increasingly complex, utilizing object-based audio systems.

Amplitude Panning utilizes Interaural Level Difference (ILD) localization cues to create a compromised sonic image placement. The signal levels are offset against each other to position individual sounds along the panoramic horizon, tricking our brains with the level differences. This "panning law" or "panning rule" only works when the listener stays within a narrow range of distances and angles from the speakers, known as the "10ms listening window." Going beyond this window can cause the Sonic Image to distort and collapse. Designing a multi-channel speaker configuration that provides adequate coverage for different audience sizes can be very challenging.

Object-based Audio (OBA) benefit:

The benefit of Object-based Audio (OBA) is that the output masters separately contain unmixed sound clips and mixing data recorded in the sound studio. All the final mixing actions are recorded in this metadata and subsequently applied at the playback sound system for the listener. This allows OBA to more accurately resemble the "Sonic Images" that the content producers created in the studio, regardless of the speaker configuration differences between the dubbing studio and the auditoriums.

What is “Circle of Confusion”?

"Circle of confusion" is the fundamental problem in the audio industry, described by legendary acoustical engineer and audio expert Floyd Toole. It explains the lack of a standardized calibration method in different stages of sound monitoring systems, i.e., manufacturing, recording studio, mixing studio, mastering studio, and consumer listening environments, leading to the sound quality being deviated from the originally intended in each stage. Listeners then judge audio equipment based on how these potentially flawed recordings sound on their own systems, creating a cycle with no fixed, objective reference point for accuracy.

The reason why we don’t have a standardized calibration method is that the sound quality cannot be replicated and be consistent in different listening environments of the sound reproduction chain, where they have their own unique reverberation patterns, in terms of our hearing perception. Due to the fact that the analyzer cannot accurately review what we actually hear.

If we have a method to keep the sound quality of the input perceptually equal to its output in each stage, we could basically break this “circuit of confusion”.

Is Cinema Dying?

The cinema industry has faced numerous disruptions over the past decades, from the rise of television to the proliferation of home video and streaming platforms. Each time, the doomsayers have proclaimed the impending demise of the movie theater experience. Yet, the cinema has endured, adapting and evolving to meet the changing demands of audiences.

History of Technological Disruptions

The advent of television in the 1950s was initially seen as a threat to the cinema industry, as people could now enjoy movies and entertainment from the comfort of their own homes. Similarly, the rise of VHS, VCD/DVD, and later, streaming services like Netflix, were all touted as the death knell for movie theaters. However, the cinema industry has proven its resilience time and time again, finding new ways to captivate audiences and remain relevant.

The Allure of Immersive Experiences

One of the key factors that has sustained the cinema's appeal is its ability to offer an unparalleled immersive experience. The combination of a large screen, surround sound, and the shared communal atmosphere creates a sensory experience that is simply unachievable in a home setting. While streaming platforms have made significant strides in video and audio quality, they still struggle to replicate the visceral impact of a cinema screening.

Personal vs. Party Atmosphere

The social aspect of cinema-going also plays a crucial role in its continued relevance. Some moviegoers prefer the personal, solitary experience of watching at home, but others cherish the energy and excitement of a shared experience with a crowd. The audience's reactions, laughter, and collective engagement can significantly enhance the viewing experience, creating a sense of community that streaming platforms struggle to emulate.

The IMAX Phenomenon

The rise of IMAX and other premium large-format (PLF) theaters has demonstrated that audiences are still willing to pay a premium for enhanced technological experiences. The larger-than-life screens, powerful sound systems, and jaw-dropping visuals have captured the imagination of moviegoers, who seek out these immersive experiences.

Netflix's Approach: Embracing Cinema

Interestingly, even Netflix, a company often portrayed as a disruptor of the cinema industry, has recognized the importance of the theatrical experience. The streaming giant has been releasing select high-profile titles in theaters, both to generate buzz and to qualify for prestigious awards. This suggests that Netflix, rather than attempting to kill the cinema industry, is using theaters as a marketing tool to promote its premium content.

Revitalizing Cinema: The Need for Innovation

While the cinema industry may be in a state of stagnation, it is not beyond saving. What the industry needs is a transformative innovation that can reinvigorate the movie-going experience and capture the imagination of modern audiences. This is where Kino Acoustics' Timbre Chart Tuning (TCT) method can play a crucial role.

The TCT method developed by Kino Acoustics is a game-changing approach to sound system calibration that can dramatically improve the audio experience in movie theaters. By precisely aligning the sound system's response with the human auditory perception, the TCT method can recreate the intended "timbre" of the original sound, ensuring that audiences hear the movie as the creators intended.

Kino Acoustics' TCT method addresses the inherent challenges of traditional sound system calibration, which often fails to account for the unique acoustic properties of each theater environment. By providing a comprehensive solution that optimizes the sound system's performance, Kino Acoustics can help cinema operators elevate the audio quality and truly immerse audiences in the cinematic experience.

Timbre Chart Tuning Method

Timbre Chart Tuning Method