src/devices/audio_latency.jd

page.title=Audio Latency
@jd:body

<!--
    Copyright 2010 The Android Open Source Project

    Licensed under the Apache License, Version 2.0 (the "License");
    you may not use this file except in compliance with the License.
    You may obtain a copy of the License at

        http://www.apache.org/licenses/LICENSE-2.0

    Unless required by applicable law or agreed to in writing, software
    distributed under the License is distributed on an "AS IS" BASIS,
    WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
    See the License for the specific language governing permissions and
    limitations under the License.
-->
<div id="qv-wrapper">
  <div id="qv">
    <h2>In this document</h2>
    <ol id="auto-toc">
    </ol>
  </div>
</div>

<p>Audio latency is the time delay as an audio signal passes through a system.
  For a complete description of audio latency for the purposes of Android
  compatibility, see <em>Section 5.5 Audio Latency</em>
  in the <a href="http://source.android.com/compatibility/index.html">Android CDD</a>.
  See <a href="latency_design.html">Design For Reduced Latency</a> for an 
  understanding of Android's audio latency-reduction efforts.
</p>

<h2 id="contributors">Contributors to Latency</h2>

<p>
  This section focuses on the contributors to output latency,
  but a similar discussion applies to input latency.
</p>
<p>
  Assuming the analog circuitry does not contribute significantly, then the major 
surface-level contributors to audio latency are the following:
</p>

<ul>
  <li>Application</li>
  <li>Total number of buffers in pipeline</li>
  <li>Size of each buffer, in frames</li>
  <li>Additional latency after the app processor, such as from a DSP</li>
</ul>

<p>
  As accurate as the above list of contributors may be, it is also misleading.
  The reason is that buffer count and buffer size are more of an
  <em>effect</em> than a <em>cause</em>.  What usually happens is that
  a given buffer scheme is implemented and tested, but during testing, an audio
  underrun is heard as a "click" or "pop."  To compensate, the
  system designer then increases buffer sizes or buffer counts.
  This has the desired result of eliminating the underruns, but it also
  has the undesired side effect of increasing latency.
</p>

<p>
  A better approach is to understand the causes of the
  underruns and then correct those.  This eliminates the
  audible artifacts and may even permit even smaller or fewer buffers
  and thus reduce latency.
</p>

<p>
  In our experience, the most common causes of underruns include:
</p>
<ul>
  <li>Linux CFS (Completely Fair Scheduler)</li>
  <li>high-priority threads with SCHED_FIFO scheduling</li>
  <li>long scheduling latency</li>
  <li>long-running interrupt handlers</li>
  <li>long interrupt disable time</li>
</ul>

<h3>Linux CFS and SCHED_FIFO scheduling</h3>
<p>
  The Linux CFS is designed to be fair to competing workloads sharing a common CPU
  resource. This fairness is represented by a per-thread <em>nice</em> parameter.
  The nice value ranges from -19 (least nice, or most CPU time allocated)
  to 20 (nicest, or least CPU time allocated). In general, all threads with a given
  nice value receive approximately equal CPU time and threads with a
  numerically lower nice value should expect to
  receive more CPU time. However, CFS is "fair" only over relatively long
  periods of observation. Over short-term observation windows,
  CFS may allocate the CPU resource in unexpected ways. For example, it
  may take the CPU away from a thread with numerically low niceness
  onto a thread with a numerically high niceness.  In the case of audio,
  this can result in an underrun.
</p>

<p>
  The obvious solution is to avoid CFS for high-performance audio
  threads. Beginning with Android 4.1, such threads now use the
  <code>SCHED_FIFO</code> scheduling policy rather than the <code>SCHED_NORMAL</code> (also called
  <code>SCHED_OTHER</code>) scheduling policy implemented by CFS.
</p>

<p>
  Though the high-performance audio threads now use <code>SCHED_FIFO</code>, they
  are still susceptible to other higher priority <code>SCHED_FIFO</code> threads.
  These are typically kernel worker threads, but there may also be a few
  non-audio user threads with policy <code>SCHED_FIFO</code>. The available <code>SCHED_FIFO</code>
  priorities range from 1 to 99.  The audio threads run at priority
  2 or 3.  This leaves priority 1 available for lower priority threads,
  and priorities 4 to 99 for higher priority threads.  We recommend 
  you use priority 1 whenever possible, and reserve priorities 4 to 99 for
  those threads that are guaranteed to complete within a bounded amount
  of time and are known to not interfere with scheduling of audio threads.
</p>

<h3>Scheduling latency</h3>
<p>
  Scheduling latency is the time between when a thread becomes
  ready to run, and when the resulting context switch completes so that the
  thread actually runs on a CPU. The shorter the latency the better, and 
  anything over two milliseconds causes problems for audio. Long scheduling
  latency is most likely to occur during mode transitions, such as
  bringing up or shutting down a CPU, switching between a security kernel
  and the normal kernel, switching from full power to low-power mode,
  or adjusting the CPU clock frequency and voltage.
</p>

<h3>Interrupts</h3>
<p>
  In many designs, CPU 0 services all external interrupts.  So a
  long-running interrupt handler may delay other interrupts, in particular
  audio direct memory access (DMA) completion interrupts. Design interrupt handlers
  to finish quickly and defer any lengthy work to a thread (preferably
  a CFS thread or <code>SCHED_FIFO</code> thread of priority 1).
</p>

<p>
  Equivalently, disabling interrupts on CPU 0 for a long period
  has the same result of delaying the servicing of audio interrupts.
  Long interrupt disable times typically happen while waiting for a kernel
  <i>spin lock</i>.  Review these spin locks to ensure that
  they are bounded.
</p>



<h2 id="measuringOutput">Measuring Output Latency</h2>

<p>
  There are several techniques available to measure output latency,
  with varying degrees of accuracy and ease of running, described below. Also
see the <a href="testing_circuit.html">Testing circuit</a> for an example test environment.
</p>

<h3>LED and oscilloscope test</h3>
<p>
This test measures latency in relation to the device's LED indicator.
If your production device does not have an LED, you can install the
  LED on a prototype form factor device. For even better accuracy
  on prototype devices with exposed circuity, connect one
  oscilloscope probe to the LED directly to bypass the light
  sensor latency.
  </p>

<p>
  If you cannot install an LED on either your production or prototype device,
  try the following workarounds:
</p>

<ul>
  <li>Use a General Purpose Input/Output (GPIO) pin for the same purpose.</li>
  <li>Use JTAG or another debugging port.</li>
  <li>Use the screen backlight. This might be risky as the
  backlight may have a non-neglible latency, and can contribute to
  an inaccurate latency reading.
  </li>
</ul>

<p>To conduct this test:</p>

<ol>
  <li>Run an app that periodically pulses the LED at
  the same time it outputs audio. 

  <p class="note"><b>Note:</b> To get useful results, it is crucial to use the correct
  APIs in the test app so that you're exercising the fast audio output path.
  See <a href="latency_design.html">Design For Reduced Latency</a> for
  background.
  </p>
  </li>
  <li>Place a light sensor next to the LED.</li>
  <li>Connect the probes of a dual-channel oscilloscope to both the wired headphone
  jack (line output) and light sensor.</li>
  <li>Use the oscilloscope to measure
  the time difference between observing the line output signal versus the light
  sensor signal.</li>
</ol>

  <p>The difference in time is the approximate audio output latency,
  assuming that the LED latency and light sensor latency are both zero.
  Typically, the LED and light sensor each have a relatively low latency
  on the order of one millisecond or less, which is sufficiently low enough
  to ignore.</p>

<h3>Larsen test</h3>
<p>
  One of the easiest latency tests is an audio feedback
  (Larsen effect) test. This provides a crude measure of combined output
  and input latency by timing an impulse response loop. This test is not very useful
  by itself because of the nature of the test, but it can be useful for calibrating 
  other tests</p>

<p>To conduct this test:</p>
<ol>
  <li>Run an app that captures audio from the microphone and immediately plays the
  captured data back over the speaker.</li>
  <li>Create a sound externally,
  such as tapping a pencil by the microphone. This noise generates a feedback loop.</li>
  <li>Measure the time between feedback pulses to get the sum of the output latency, input latency, and application overhead.</li>
</ol>

  <p>This method does not break down the
  component times, which is important when the output latency
  and input latency are independent. So this method is not recommended for measuring output latency, but might be useful
  to help measure output latency.</p>

<h2 id="measuringInput">Measuring Input Latency</h2>

<p>
  Input latency is more difficult to measure than output latency. The following
  tests might help.
</p>

<p>
One approach is to first determine the output latency
  using the LED and oscilloscope method and then use
  the audio feedback (Larsen) test to determine the sum of output
  latency and input latency. The difference between these two
  measurements is the input latency.
</p>

<p>
  Another technique is to use a GPIO pin on a prototype device.
  Externally, pulse a GPIO input at the same time that you present
  an audio signal to the device.  Run an app that compares the
  difference in arrival times of the GPIO signal and audio data.
</p>

<h2 id="reducing">Reducing Latency</h2>

<p>To achieve low audio latency, pay special attention throughout the
system to scheduling, interrupt handling, power management, and device
driver design. Your goal is to prevent any part of the platform from
blocking a <code>SCHED_FIFO</code> audio thread for more than a couple
of milliseconds. By adopting such a systematic approach, you can reduce
audio latency and get the side benefit of more predictable performance
overall.
</p>


 <p>
  Audio underruns, when they do occur, are often detectable only under certain
  conditions or only at the transitions. Try stressing the system by launching
  new apps and scrolling quickly through various displays. But be aware
  that some test conditions are so stressful as to be beyond the design
  goals. For example, taking a bugreport puts such enormous load on the
  system that it may be acceptable to have an underrun in that case.
</p>

<p>
  When testing for underruns:
</p>
  <ul>
  <li>Configure any DSP after the app processor so that it adds
  minimal latency.</li>
  <li>Run tests under different conditions
  such as having the screen on or off, USB plugged in or unplugged,
  WiFi on or off, Bluetooth on or off, and telephony and data radios
  on or off.</li>
  <li>Select relatively quiet music that you're very familiar with, and which is easy
  to hear underruns in.</li>
  <li>Use wired headphones for extra sensitivity.</li>
  <li>Give yourself breaks so that you don't experience "ear fatigue."</li>
  </ul>

<p>
  Once you find the underlying causes of underruns, reduce
  the buffer counts and sizes to take advantage of this.
  The eager approach of reducing buffer counts and sizes <i>before</i>
  analyzing underruns and fixing the causes of underruns only
  results in frustration.
</p>

<h3 id="tools">Tools</h3>
<p>
  <code>systrace</code> is an excellent general-purpose tool
  for diagnosing system-level performance glitches.
</p>

<p>
  The output of <code>dumpsys media.audio_flinger</code> also contains a
  useful section called "simple moving statistics." This has a summary
  of the variability of elapsed times for each audio mix and I/O cycle.
  Ideally, all the time measurements should be about equal to the mean or
  nominal cycle time. If you see a very low minimum or high maximum, this is an
  indication of a problem, likely a high scheduling latency or interrupt
  disable time. The <i>tail</i> part of the output is especially helpful,
  as it highlights the variability beyond +/- 3 standard deviations.
</p>