Sample rate refers to the amount of samples which are carried per second. The higher the sample rate, the lower the latency. Higher samples rates however also put additional stress on the CPU.
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Potential contributors to latency in an audio system include analog-to-digital conversion, buffering, digital signal processing, transmission time, digital-to-analog conversion and the speed of sound in the transmission medium.
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BUFFER SIZE / SAMPLE RATE = LATENCY
Sample rate is how many times per second that a sample is captured. Sample rate also determines the highest frequency that can be accurately captured. Higher sample rates allow for capturing higher frequencies.
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The more samples you take – known as the ‘sample rate’ – the more closely the final digital file will resemble the original. A higher sample rate tends to deliver a better-quality audio reproduction. Sample rates are usually measured per second, using kilohertz (kHz) or cycles per second.
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How to fix microphone latency:
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How to reduce audio interface latency
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The most common reasons why latency occurs are the audio driver(s) and hardware you’re using, and also the buffer size and sample rate.
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Drawbacks of High Sample Rates
In theory, a higher sample rate will only capture frequencies at extremely high and low ends of the spectrum where listeners can’t even hear them. This means you’re spending more and using more space for music that doesn’t have a noticeable improvement in sound.
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If the rate of samples is too high the system may not be able to process them fast enough – it runs out of processing time.
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You should match whatever rate the file is, if you play a 16bit 44100hz file at 24bit 48000hz you will get lower quality sound due to upscaling. Same goes for vice versa due to downscaling. Anywho, if the file rate is unknown I would stick to 16bit 44100hz as that is cd quality audio and will be more common.
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Higher sample rate means more data into the computer and larger files. How can that also mean it’s faster? Because input and output buffers are shorter with faster SR.
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A latency time of around 10ms or less usually means that it won’t affect the recording process. Above 10 ms, the effect starts to become noticeable.
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While some keyboardists claim to hear a 5ms discrepancy in their performances, the vast majority of musicians are unlikely to worry about 10ms, and many should find a latency of 23ms or more perfectly acceptable with most sounds, especially pads with longer attacks.
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What Causes Latency? In most situations, latency is caused by your internet network hardware, your remote server’s location and connection, and the internet routers that are located between your server and your online gaming device, smartphone, tablet or other internet device.
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What causes Internet latency? One of the principal causes of network latency is distance, specifically the distance between client devices making requests and the servers responding to those requests.
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Latency is affected by several factors: distance, propagation delay, internet connection type, website content, Wi-Fi, and your router. Some of these factors are fixable, while others are just part of everyone’s online experience.
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The higher the sampling rate, the more accurate the sound wave representation will be. Lower sampling rates mean fewer samples per second. With less audio data, the audio representation will be approximate, to some extent. The most common sampling rate values are 44.1 kHz and 48 kHz.
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Is a higher audio sample rate better? In theory, it’s not a bad idea to work at a higher audio sample rate, like 176.4 kHz or 192 kHz. The files will be larger, but it can be nice to maximize the sound quality until the final bounce. In the end, however, the audio will likely be converted to either 44.1 kHz or 48 kHz.
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Very large samples tend to transform small differences into statistically significant differences – even when they are clinically insignificant. As a result, both researchers and clinicians are misguided, which may lead to failure in treatment decisions.
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Analog recordings such as vinyl and tape have long been considered the “gold standard” for sound quality among audiophiles. High-resolution audio achieves this supposed high quality by offering music files encoded with 24-bit depth and a sample rate of 192 kHz (24/192) as well as other ranges including 24/96.
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For mastering, 96kHz or even archival mastering at 192kHz is usually a good idea. Regardless, recording at 44.1 or 48kHz through a high-quality modern audio interface will give you excellent results, depending on the situation, very similar to what you’d get at higher rates.
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In the end, 44.1 kHz, 88.2 kHz, and 176.4 kHz sample rates remain the preferred options for studio recording when outputting to a CD. However, if your audio is to be used in a video or delivered on a digital format, 48 kHz and its derivatives may provide the better option.
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Latency’s effects depend on observers, but most will perceive obvious latency around 100 – 120 milliseconds. Communications will start to break down around 250 – 300ms.
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In direct monitoring (also known as zero-latency monitoring), the audio signal is rerouted back to the monitors or headphones without processing. This means you will hear your voice without any effects or processing and with no perceivable delay.
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While we can’t generally hear the effects of latency until they are around 15-30 milliseconds (ms), performers can begin to feel them at around 5-10ms. At 7ms, latency starts to mess with our ability to play or sing on top of or behind the beat. Sound starts to feel sluggish at 10ms. These are generalities, of course.
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