Jitter is time distortions of recording/playback of digital audio signal due clock deviation. More than one digital audio system owner worry about it. Other people consider digital audio jitter issue as myth, that is inaudible. Read about reasons, estimation, suppression, can you hear jitter or not and other in the article.
Digital-analog converter (DAC) transform sample sequence (digital values) to analog voltage level sequence. In the ideal case, same time distance between the samples should be provided. The distance is defined by clocking. However, clocking in not ideal and restored signal is distorted like it is shown at the picture.
Jitter is deviation of time between samples (deviation of sampling rate).
Jitter time distortions of playing back digital signal
At upper left part of the picture we can see original musical signal (green) captured into digital form.
At the bottom left part of the picture we can see restoration of the captured digital signal back to analog form.
The samples (vertical lines with dot) have unstable time position (jitter) at horizontal axis. And restored signal is distorted. We can see it at right part of the picture.
Jitter is non-linear distortions. Theoretically, we can artifically boost jitter deviation and listen more noise and artefacts.
In real life, jitter always impact to analog signal at DAC output. But it's too small for modern systems. I think, it is impossibly or almost impossibly to listen jitter. Because jitter distortions compete with quantization noise, non-linear distortions and own noise of electronic components. I'm not sure, that we can separate real-system-jitter noise and rest noise/distortions.
Jitter is sample clock issue (clock deviation).
Jitter appear in moment when digital signal in line is transformed to binary sequence.
How the jitter appear
[bit value (0/1) detection]
To transmit music data thru line (cable), bit sequence is converted to electrical form. Coding in the electrical form may be implemented different ways (voltage level values or other).
Here we consider simple amplitude coding: signal voltage level above threshold (dot line at the picture) is binary value 1, below is binary value 0.
The line transmitter device convert binary data to electrical levels. The line receiver device convert electrical levels back to binary sequence.
And clock reference moments is the points, where voltage level get higher the threshold.
The line transmitter generate signal close to square.
In the line signal "lose" form due noise, frequency and non-linear distortions.
Thus clock reference moments, detecting in the line receiver, may be offset from initial time position (time deviation at the picture).
Below we will consider where the jitter penetrate to the audio system.
Jitter is non linear distortions. To measure jitter need put pure sine to input of a learned system.
Scheme of the jitter measurement
At the system output we check artifacts (harmonics) and noise.
Jitter spectrum
Jitter products (artifacts and noise) depend on the input signal. To check the dependency we can take spectrum for different input signal level and frequency.
At the picture is not real spectrum. It is only illustration for better understanding.
We can measure:
Measurements #2 and #3 allow to separate errors of system parts.
Jitter is deviation time between samples. Delay is shifting of full signal waveform with keeping time distance between samples.
Latency is delay for processing inside audio device (read below about FIFO buffer).
Above we considered, that clock deviation is effect of digital signal distortions.
There are several factors that impact to clock deviation:
Clock generator have electronic elements, that define frequency and its deviation.
Changing of power DC voltage can cause frequency deviations of the clock generator.
Power supply unit can generate noise into electrical lines. This noise can modulate the generated clock impulses.
Modified clock signal cause the time deviation (see the picture).
Noise, that impact to digital audio signal line, penetrate from electrical circuits and the air.
Unfortunately, jitter is not pure DAC issue. When we use recorded stuff, we have jittered sequence of digital samples.
To understand fully jitter issue need to learn full recording-playback system.
Jitter into full audio system
Capturing of music is periodical measurements of analog signal. If the periods will vary, it cause time distortions in recorded audio samples.
Jitter of audio recording.
Let's compare capturing with jitter and without
In the upper left part of the picture signal captured without jitter.
In the lower left part of the picture signal captured with jitter.
We can see that samples for both these cases have different values.
I.e. after restoration to analog form (playback) signal, captured with and without jitter, have different waveforms of the analog signal.
If music was recorded with jitter errors we can't compensate it further. Because the jitter is random.
As rule, in a recording studios use professional apparatus, including dedicated clock sources (Word Clock [1], as example). Therefore, engineers in the studios try maximally decrease losses due bad clock.
Let's look to point #2 (digital signal pass thru the digital audio system). When signal is placed into digital domain (signal in digital form) jitter is not matter. Because time scores for binary signal form have pure mathematical values with infinite precision.
Any delays in processing between samples are not matter for restored signal.
However, processing delays into digital domain can cause real-time interruptions of data stream, that feed DAC. But it is not jitter issue. Read details below.
Now let's look to point #3 (signal come to DAC from pure digital part of the audio system).
Noise, that cause jitter, penetrate to digital signal several ways from:
However, FIFO buffers provide "jitter isolation" between any "jittered" segment.
FIFO ("first input, first output") kind of buffer (array of number samples) when samples out from buffer on a first-come.
In the article putting and getting of samples to/from buffer are asynchronous. Clocks of buffer writing and reading are independent.
FIFO buffer cause time delay between sample input and output (latency). Latency is measured in the seconds, milliseconds, microseconds.
Short buffers cause small delay value. Short latency is important for real-time audio systems, used for music live performance and production. For home audio big latency value almost is not matter, except real-time sound adjusting.
FIFO buffer with asynchronous writing (input) and reading (output).
Sample move thru buffer step-by-step
Jitter before FIFO-buffer don't impact to clock after FIFO. Because the buffer is asynchronous.
Therefore, there are no reasons to worry about jitter before FIFO into DAC for scheme considered at the picture above.
For an audio system, before suggesting of effective jitter suppression, learning of the system scheme is recommended.
In most cases, asynchronous FIFO buffer allow to fix interruptions in a binary data stream. Of course, significant interruptions can't be fixed by FIFO. The non-fixed interruptions cause pauses, clicks or other sound damages. It happens when the buffer is empty and no binary data ready for conversion to analog form.
DAC clock synchronization have 3 options:
Below we consider cases for DAC with internal FIFO of input audio data.
Synchronization by digital interface have jitter sources:
Synchronization by DAC's internal clock generator have primary jitter sources:
Synchronization by external dedicated clock generator have jitter sources:
For home applications I'd recommend to use DAC's internal clock generator as simplest way. It may be chosen in the DAC's settings.
However, it is not guarantee the best result. Each setup should be analyzed (measured) to choose the minimal jitter configuration.
Yuri Korzunov,
Audiophile Inventory developer, Google+
References
Read about audio issues
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