[ardour-manual] / include / latency-and-latency-compensation.html

<p>
  <a
  href="http://en.wikipedia.org/wiki/Latency_%28audio%29"><dfn>Latency</dfn></a> 
  is a system's reaction time to a given stimulus. There are many factors that 
  contribute to the total latency of a system. In order to achieve exact time 
  synchronization all sources of latency need to be taken into account and 
  compensated for.
</p>

<h2>Sources of Latency</h2>

<h3>Sound propagation through the air</h3>
<p>
  Since sound is a mechanical perturbation in a fluid, it travels at 
  comparatively slow <a href="http://en.wikipedia.org/wiki/Speed_of_sound">speed</a> 
  of about 340 m/s. As a consequence, your acoustic guitar or piano has a 
  latency of about 1&ndash;2 ms, due to the propagation time of the sound 
  between your instrument and your ear. 
</p>
<h3>Digital-to-Analog and Analog-to-Digital conversion</h3>
<p>
  Electric signals travel quite fast (on the order of the speed of light), 
  so their propagation time is negligible in this context. But the conversions 
  between the analog and digital domain take a comparatively long time to perform, 
  so their contribution to the total latency may be considerable on
  otherwise very low-latency systems. Conversion delay is usually below 1&nbsp;ms.
</p>
<h3>Digital Signal Processing</h3>
<p>
  Digital processors tend to process audio in chunks, and the size of that chunk 
  depends on the needs of the algorithm and performance/cost considerations. 
  This is usually the main cause of latency when you use a computer and one you 
  can try to predict and optimize.
</p>
<h3>Computer I/O Architecture</h3>
<p>
  A computer is a general purpose processor, not a digital audio processor. 
  This means our audio data has to jump a lot of fences in its path from the 
  outside to the CPU and back, contending in the process with some other parts 
  of the system vying for the same resources (CPU time, bus bandwidth, etc.) 
</p>

<h2>The Latency chain</h2>

<img src="/images/latency-chain.png"  title="Latency chain" alt="Latency chain" />
<p>
  <em>Figure: Latency chain.</em> 
  The numbers are an example for a typical PC. With professional gear and an 
  optimized system the total roundtrip latency is usually lower. The important 
  point is that latency is always additive and a sum of many independent factors.
</p>

<p>
  Processing latency is usually divided into <dfn>capture latency</dfn> (the time 
  it takes for the digitized audio to be available for digital processing, usually 
  one audio period), and <dfn>playback latency</dfn> (the time it takes for
  In practice, the combination of both matters. It is called <dfn>roundtrip 
  latency</dfn>: the time necessary for a certain audio event to be captured, 
  processed and played back.
</p>
<p class="note">
  It is important to note that processing latency in a jackd is a matter of
  choice. It can be lowered within the limits imposed by the hardware (audio 
  device, CPU and bus speed) and audio driver. Lower latencies increase the 
  load on the system because it needs to process the audio in smaller chunks 
  which arrive much more frequently. The lower the latency, the more likely 
  the system will fail to meet its processing deadline and the dreaded
  <dfn>xrun</dfn> (short for buffer over- or under-run) will make its 
  appearance more often, leaving its merry trail of clicks, pops and crackles.
</p>

<p>
  The digital I/O latency is usually negligible for integrated or 
  <abbr title="Periphal Component Interface">PCI</abbr> audio devices, but 
  for USB or FireWire interfaces the bus clocking and buffering can add some 
  milliseconds.
</p>


<h2>Low Latency usecases</h2>
<p>
  Low latency is <strong>not</strong> always a feature you want to have. It 
  comes with a couple of drawbacks: the most prominent is increased power 
  consumption because the CPU needs to process many small chunks of audio data,
  it is constantly active and can not enter power-saving mode (think fan-noise). 
  Since each application that is part of the signal chain must run in every 
  audio cycle, low-latency systems will undergo<dfn>context switches</dfn> 
  between applications more often, which incur a significant overhead.  
  This results in a much higher system load and an increased chance of xruns. 
</p>
<p>
  For a few applications, low latency is critical:
</p>
<h3>Playing virtual instruments</h3>
<p>
  A large delay between the pressing of the keys and the sound the instrument 
  produces will throw-off the timing of most instrumentalists (save church 
  organists, whom we believe to be awesome latency-compensation organic systems.)
</p>
<h3>Software audio monitoring</h3>
<p>
  If a singer is hearing her own voice through two different paths, her head 
  bones and headphones, even small latencies can be very disturbing and
  manifest as a tinny, irritating sound.
</p>
<h3>Live effects</h3>
<p>
  Low latency is important when using the computer as an effect rack for
  inline effects such as compression or EQ. For reverbs, slightly higher
  latency might be tolerable, if the direct sound is not routed through the
  computer. 
</p>
<h3>Live mixing</h3> 
<p>
  Some sound engineers use a computer for mixing live performances. 
  Basically that is a combination of the above: monitoring on stage, 
  effects processing and EQ. 
</p>
<p>
  In many other cases, such as playback, recording, overdubbing, mixing, 
  mastering, etc. latency is not important, since it can easily be 
  compensated for.<br />
  To explain that statement: During mixing or mastering you don&#039;t care 
  if it takes 10ms or 100ms between the instant you press the play button
  and sound coming from the speaker. The same is true when recording with a count in.
</p>

<h2>Latency compensation</h2>
<p>
  During tracking it is important that the sound that is currently being 
  played back is internally aligned with the sound that is being recorded.
</p>
<p>
  This is where latency-compensation comes into play. There are two ways to 
  compensate for latency in a DAW, <dfn>read-ahead</dfn> and
  <dfn>write-behind</dfn>. The DAW starts playing a bit early (relative to 
  the playhead), so that when the sound arrives at the speakers a short time 
  later, it is exactly aligned with the material that is being recorded.
  Since we know that play-back has latency, the incoming audio can be delayed 
  by the same amount to line things up again.
</p>
<p>
  As you may see, the second approach is prone to various implementation 
  issues regarding timecode and transport synchronization. Ardour uses read-ahead 
  to compensate for latency. The time displayed in the Ardour clock corresponds 
  to the audio-signal that you hear on the speakers (and is not where Ardour 
  reads files from disk).
</p>
<p>
  As a side note, this is also one of the reasons why many projects start at 
  timecode <samp>01:00:00:00</samp>. When compensating for output latency the 
  DAW will need to read data from before the start of the session, so that the 
  audio arrives in time at the output when the timecode hits <samp>01:00:00:00</samp>. 
  Ardour3 does handle the case of <samp>00:00:00:00</samp> properly but not all 
  systems/software/hardware that you may inter-operate with may behave the same.
</p>

<h2>Latency Compensation And Clock Sync</h2>

<p>
  To achieve sample accurate timecode synchronization, the latency introduced 
  by the audio setup needs to be known and compensated for.
</p>
<p>
  In order to compensate for latency, JACK or JACK applications need to know 
  exactly how long a certain signal needs to be read-ahead or delayed:
</p>
<img src="/images/jack-latency-excerpt.png"  title="Jack Latency Compensation" alt="Jack Latency Compensation" />
<p>
  <em>Figure: Jack Latency Compensation.</em>  
</p>
<p>
  In the figure above, clients A and B need to be able to answer the following 
  two questions:
</p>
<ul>
  <li>
    How long has it been since the data read from port Ai or Bi arrived at the
    edge of the JACK graph (capture)?
  </li>
  <li>
    How long will it be until the data writen to port Ao or Bo arrives at the
    edge of the JACK graph (playback)?
  </li>
</ul>

<p>
  JACK features an <abbr title="Application Programming Interface">API</abbr> 
  that allows applications to determine the answers to above questions. 
  However JACK can not know about the additional latency that is introduced 
  by the computer architecture, operating system and soundcard. These values 
  can be specified by the JACK command line parameters <kbd class="input">-I</kbd> 
  and <kbd class="input">-O</kbd> and vary from system 
  to system but are constant on each. On a general purpose computer system 
  the only way to accurately learn about the total (additional) latency is to 
  measure it.
</p>


<h2>Calibrating JACK Latency</h2>
<p>
  Linux DSP guru Fons Adriaensen wrote a tool called <dfn>jack_delay</dfn> 
  to accurately measure the roundtrip latency of a closed loop audio chain, 
  with sub-sample accuracy. JACK itself includes a variant of this tool 
  called <dfn>jack_iodelay</dfn>.
</p>
<p>
  Jack_iodelay allows you to measure the total latency of the system, 
  subtracts the known latency of JACK itself and suggests values for 
  jackd's audio-backend parameters.
</p>
<p>
  jack_[io]delay works by emitting some rather annoying tones, capturing 
  them again after a round trip through the whole chain, and measuring the 
  difference in phase so it can estimate with great accuracy the time taken. 
</p>
<p>
  You can close the loop in a number of ways:
</p>
<ul>
  <li>
    Putting a speaker close to a microphone. This is rarely done, as air 
    propagation latency is well known so there is no need to measure it.
  </li>
  <li>
    Connecting the output of your audio interface to its input using a 
    patch cable. This can be an analog or a digital loop, depending on 
    the nature of the input/output you use. A digital loop will not factor 
    in the <abbr title="Analog to Digital, Digital to Analog">AD/DA</abbr> 
    converter latency.
  </li>
</ul>
<p>
  Once you have closed the loop you have to:
</p>
<ol>
  <li>Launch jackd with the configuration you want to test.</li>
  <li>Launch <kbd class="input">jack_delay</kbd> on the commandline.</li>
  <li>Make the appropriate connections between your jack ports so the loop is closed.</li>
  <li>Adjust the playback and capture levels in your mixer.</li>
</ol>