4 href="http://en.wikipedia.org/wiki/Latency_%28audio%29"><dfn>Latency</dfn></a>
5 is a system's reaction time to a given stimulus. There are many factors that
6 contribute to the total latency of a system. In order to achieve exact time
7 synchronization all sources of latency need to be taken into account and
11 <h2>Sources of Latency</h2>
13 <h3>Sound propagation through the air</h3>
15 Since sound is a mechanical perturbation in a fluid, it travels at
16 comparatively slow <a href="http://en.wikipedia.org/wiki/Speed_of_sound">speed</a>
17 of about 340 m/s. As a consequence, your acoustic guitar or piano has a
18 latency of about 1–2 ms, due to the propagation time of the sound
19 between your instrument and your ear.
21 <h3>Digital-to-Analog and Analog-to-Digital conversion</h3>
23 Electric signals travel quite fast (on the order of the speed of light),
24 so their propagation time is negligible in this context. But the conversions
25 between the analog and digital domain take a comparatively long time to perform,
26 so their contribution to the total latency may be considerable on
27 otherwise very low-latency systems. Conversion delay is usually below 1 ms.
29 <h3>Digital Signal Processing</h3>
31 Digital processors tend to process audio in chunks, and the size of that chunk
32 depends on the needs of the algorithm and performance/cost considerations.
33 This is usually the main cause of latency when you use a computer and one you
34 can try to predict and optimize.
36 <h3>Computer I/O Architecture</h3>
38 A computer is a general purpose processor, not a digital audio processor.
39 This means our audio data has to jump a lot of fences in its path from the
40 outside to the CPU and back, contending in the process with some other parts
41 of the system vying for the same resources (CPU time, bus bandwidth, etc.)
44 <h2>The Latency chain</h2>
46 <img src="/images/latency-chain.png" title="Latency chain" alt="Latency chain" />
48 <em>Figure: Latency chain.</em>
49 The numbers are an example for a typical PC. With professional gear and an
50 optimized system the total roundtrip latency is usually lower. The important
51 point is that latency is always additive and a sum of many independent factors.
55 Processing latency is usually divided into <dfn>capture latency</dfn> (the time
56 it takes for the digitized audio to be available for digital processing, usually
57 one audio period), and <dfn>playback latency</dfn> (the time it takes for
58 In practice, the combination of both matters. It is called <dfn>roundtrip
59 latency</dfn>: the time necessary for a certain audio event to be captured,
60 processed and played back.
63 It is important to note that processing latency in a jackd is a matter of
64 choice. It can be lowered within the limits imposed by the hardware (audio
65 device, CPU and bus speed) and audio driver. Lower latencies increase the
66 load on the system because it needs to process the audio in smaller chunks
67 which arrive much more frequently. The lower the latency, the more likely
68 the system will fail to meet its processing deadline and the dreaded
69 <dfn>xrun</dfn> (short for buffer over- or under-run) will make its
70 appearance more often, leaving its merry trail of clicks, pops and crackles.
74 The digital I/O latency is usually negligible for integrated or
75 <abbr title="Periphal Component Interface">PCI</abbr> audio devices, but
76 for USB or FireWire interfaces the bus clocking and buffering can add some
81 <h2>Low Latency usecases</h2>
83 Low latency is <strong>not</strong> always a feature you want to have. It
84 comes with a couple of drawbacks: the most prominent is increased power
85 consumption because the CPU needs to process many small chunks of audio data,
86 it is constantly active and can not enter power-saving mode (think fan-noise).
87 Since each application that is part of the signal chain must run in every
88 audio cycle, low-latency systems will undergo<dfn>context switches</dfn>
89 between applications more often, which incur a significant overhead.
90 This results in a much higher system load and an increased chance of xruns.
93 For a few applications, low latency is critical:
95 <h3>Playing virtual instruments</h3>
97 A large delay between the pressing of the keys and the sound the instrument
98 produces will throw-off the timing of most instrumentalists (save church
99 organists, whom we believe to be awesome latency-compensation organic systems.)
101 <h3>Software audio monitoring</h3>
103 If a singer is hearing her own voice through two different paths, her head
104 bones and headphones, even small latencies can be very disturbing and
105 manifest as a tinny, irritating sound.
107 <h3>Live effects</h3>
109 Low latency is important when using the computer as an effect rack for
110 inline effects such as compression or EQ. For reverbs, slightly higher
111 latency might be tolerable, if the direct sound is not routed through the
116 Some sound engineers use a computer for mixing live performances.
117 Basically that is a combination of the above: monitoring on stage,
118 effects processing and EQ.
121 In many other cases, such as playback, recording, overdubbing, mixing,
122 mastering, etc. latency is not important, since it can easily be
124 To explain that statement: During mixing or mastering you don't care
125 if it takes 10ms or 100ms between the instant you press the play button
126 and sound coming from the speaker. The same is true when recording with a count in.
129 <h2>Latency compensation</h2>
131 During tracking it is important that the sound that is currently being
132 played back is internally aligned with the sound that is being recorded.
135 This is where latency-compensation comes into play. There are two ways to
136 compensate for latency in a DAW, <dfn>read-ahead</dfn> and
137 <dfn>write-behind</dfn>. The DAW starts playing a bit early (relative to
138 the playhead), so that when the sound arrives at the speakers a short time
139 later, it is exactly aligned with the material that is being recorded.
140 Since we know that play-back has latency, the incoming audio can be delayed
141 by the same amount to line things up again.
144 As you may see, the second approach is prone to various implementation
145 issues regarding timecode and transport synchronization. Ardour uses read-ahead
146 to compensate for latency. The time displayed in the Ardour clock corresponds
147 to the audio-signal that you hear on the speakers (and is not where Ardour
148 reads files from disk).
151 As a side note, this is also one of the reasons why many projects start at
152 timecode <samp>01:00:00:00</samp>. When compensating for output latency the
153 DAW will need to read data from before the start of the session, so that the
154 audio arrives in time at the output when the timecode hits <samp>01:00:00:00</samp>.
155 Ardour3 does handle the case of <samp>00:00:00:00</samp> properly but not all
156 systems/software/hardware that you may inter-operate with may behave the same.
159 <h2>Latency Compensation And Clock Sync</h2>
162 To achieve sample accurate timecode synchronization, the latency introduced
163 by the audio setup needs to be known and compensated for.
166 In order to compensate for latency, JACK or JACK applications need to know
167 exactly how long a certain signal needs to be read-ahead or delayed:
169 <img src="/images/jack-latency-excerpt.png" title="Jack Latency Compensation" alt="Jack Latency Compensation" />
171 <em>Figure: Jack Latency Compensation.</em>
174 In the figure above, clients A and B need to be able to answer the following
179 How long has it been since the data read from port Ai or Bi arrived at the
180 edge of the JACK graph (capture)?
183 How long will it be until the data writen to port Ao or Bo arrives at the
184 edge of the JACK graph (playback)?
189 JACK features an <abbr title="Application Programming Interface">API</abbr>
190 that allows applications to determine the answers to above questions.
191 However JACK can not know about the additional latency that is introduced
192 by the computer architecture, operating system and soundcard. These values
193 can be specified by the JACK command line parameters <kbd class="input">-I</kbd>
194 and <kbd class="input">-O</kbd> and vary from system
195 to system but are constant on each. On a general purpose computer system
196 the only way to accurately learn about the total (additional) latency is to
201 <h2>Calibrating JACK Latency</h2>
203 Linux DSP guru Fons Adriaensen wrote a tool called <dfn>jack_delay</dfn>
204 to accurately measure the roundtrip latency of a closed loop audio chain,
205 with sub-sample accuracy. JACK itself includes a variant of this tool
206 called <dfn>jack_iodelay</dfn>.
209 Jack_iodelay allows you to measure the total latency of the system,
210 subtracts the known latency of JACK itself and suggests values for
211 jackd's audio-backend parameters.
214 jack_[io]delay works by emitting some rather annoying tones, capturing
215 them again after a round trip through the whole chain, and measuring the
216 difference in phase so it can estimate with great accuracy the time taken.
219 You can close the loop in a number of ways:
223 Putting a speaker close to a microphone. This is rarely done, as air
224 propagation latency is well known so there is no need to measure it.
227 Connecting the output of your audio interface to its input using a
228 patch cable. This can be an analog or a digital loop, depending on
229 the nature of the input/output you use. A digital loop will not factor
230 in the <abbr title="Analog to Digital, Digital to Analog">AD/DA</abbr>
235 Once you have closed the loop you have to:
238 <li>Launch jackd with the configuration you want to test.</li>
239 <li>Launch <kbd class="input">jack_delay</kbd> on the commandline.</li>
240 <li>Make the appropriate connections between your jack ports so the loop is closed.</li>
241 <li>Adjust the playback and capture levels in your mixer.</li>