docs/html/ndk/guides/audio/input-latency.jd

page.title=Audio Input Latency
@jd:body

<div id="qv-wrapper">
    <div id="qv">
      <h2>On this page</h2>

      <ol>
        <li><a href="#check-list">Checklist</a></li>
        <li><a href="#ways">Ways to Reduce Audio Input Latency</a></li>
        <li><a href="#avoid">What to Avoid</a></li>
      </ol>
    </div>
  </div>


<p>This page provides guidelines to help you reduce audio input latency when recording with a
built-in microphone or an external headset microphone.</p>

<h2 id="check-list">Checklist</h2>

<p>Here are a few important prerequisites:</p>

<ul>
  <li>You must use the Android-specific implementation of the
  <a class="external-link" href="https://www.khronos.org/opensles/">OpenSL ES™</a> API.

  <li>If you haven't already done so, download and install the
  <a href="{@docRoot}tools/sdk/ndk/index.html">Android NDK</a>.</li>

  <li>Many of the same requirements for low-latency audio output also apply to low-latency input,
  so read the requirements for low-latency output in
  <a href="{@docRoot}ndk/guides/audio/output-latency.html">Audio Output Latency</a>.</li>
</ul>

<h2 id="ways">Ways to Reduce Audio Input Latency</h2>

<p>The following are some methods to help ensure low audio input latency:

<ul>
  <li>Suggest to your users, if your app relies on low-latency audio, that they use a headset
(for example, by displaying a <em>Best with headphones</em> screen on first run). Note
that just using the headset doesn’t guarantee the lowest possible latency. You may need to
perform other steps to remove any unwanted signal processing from the audio path, such as by
using the <a href="http://developer.android.com/reference/android/media/MediaRecorder.AudioSource.html#VOICE_RECOGNITION">
VOICE_RECOGNITION</a> preset when recording.</li>

  <li>It's difficult to test audio input and output latency in isolation. The best solution to
determine the lowest possible audio input latency is to measure round-trip audio and divide
by two.</li>
 <li> Be prepared to handle nominal sample rates of 44,100 and 48,000 Hz as reported by
<a href="{@docRoot}reference/android/media/AudioManager.html#getProperty(java.lang.String)">
getProperty(String)</a> for
<a href="{@docRoot}reference/android/media/AudioManager.html#PROPERTY_OUTPUT_SAMPLE_RATE">
PROPERTY_OUTPUT_SAMPLE_RATE</a>. Other sample rates are possible, but rare.</li>

  <li>Be prepared to handle the buffer size reported by
<a href="{@docRoot}reference/android/media/AudioManager.html#getProperty(java.lang.String)">
getProperty(String)</a> for
<a href="{@docRoot}reference/android/media/AudioManager.html#PROPERTY_OUTPUT_FRAMES_PER_BUFFER">
PROPERTY_OUTPUT_FRAMES_PER_BUFFER</a>. Typical buffer sizes include 96, 128, 160, 192, 240, 256,
or 512 frames, but other values are possible.</li>
</ul>

<h2 id="avoid">What to Avoid</h2>

<p>Be sure to take these things into account to help avoid latency issues:</p>

<ul>
  <li>Don’t assume that the speakers and microphones used in mobile devices generally have good
acoustics. Due to their small size, the acoustics are generally poor so signal processing is
added to improve the sound quality. This signal processing introduces latency.</li>

  <li>Don't assume that your input and output callbacks are synchronized. For simultaneous input
and output, separate buffer queue completion handlers are used for each side. There is no
guarantee of the relative order of these callbacks or the synchronization of the audio clocks,
even when both sides use the same sample rate. Your application should buffer the data with
proper buffer synchronization.</li>

  <li>Don't assume that the actual sample rate exactly matches the nominal sample rate. For
example, if the nominal sample rate is 48,000 Hz, it is normal for the audio clock to advance
at a slightly different rate than the operating system {@code CLOCK_MONOTONIC}. This is because
the audio and system clocks may derive from different crystals.</li>

  <li>Don't assume that the actual playback sample rate exactly matches the actual capture sample
rate, especially if the endpoints are on separate paths. For example, if you are capturing from
the on-device microphone at 48,000 Hz nominal sample rate, and playing on USB audio
at 48,000 Hz nominal sample rate, the actual sample rates are likely to be slightly different
from each other.</li>
</ul>

<p>A consequence of potentially independent audio clocks is the need for asynchronous sample rate
conversion. A simple (though not ideal for audio quality) technique for asynchronous sample rate
conversion is to duplicate or drop samples as needed near a zero-crossing point. More
sophisticated conversions are possible.</p>