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Latency is the time it takes for a signal to travel through a system. These are the commontypes of latency related to audio apps:
- Audio output latency is the time between an audio sample being generated by anapp and the sample being played through the headphone jack or built-in speaker.
- Audio input latency is the time between an audio signal being received by adevice’s audio input, such as the microphone, and that same audio data being available to anapp.
- Round-trip latency is the sum of input latency, app processing time, and output latency.
- Touch latency is the time between a user touching the screen and thattouch event being received by an app.
- Warmup latency is the time it takes to start up the audio pipeline the first time data is enqueued in a buffer.
This page describes how to develop your audio app with low-latency input and output, and how to avoidwarmup latency.
Measure latency
It is difficult to measure audio input and output latency in isolation since it requires knowing exactlywhen the first sample is sent into the audio path (although this can be done using alight testing circuit and an oscilloscope). If you know the round-trip audio latency, you canuse the rough rule of thumb: audio input (and output) latency is half the round-trip audio latencyover paths without signal processing.
Round-trip audio latency varies greatly depending on device model andAndroid build. You can get a rough idea of round-triplatency for Nexus devices by reading thepublished measurements.
You can measure round-trip audio latency by creating an app that generates an audio signal,listens for that signal, and measures the time between sending it and receiving it.Alternatively, you can install thislatency testing app. This performs a round-trip latency test using theLarsen test. You can alsoview the source code for the latency testing app.
Since the lowest latency is achieved over audio paths with minimal signal processing, you mayalso want to use anAudio Loopback Dongle, which allows the test to be run over the headset connector.
Best practices to minimize latency
Validate audio performance
Enqueue 1 0 1 – Fast Light Weight Audio Player Manual
The Android Compatibility Definition Document (CDD) enumerates the hardware and softwarerequirements of a compatible Android device.See Android Compatibility for more information on the overall compatibility program, andCDD for the actual CDD document.
In the CDD, round-trip latency is specified as 20 ms or lower (even though musiciansgenerally require 10 ms). This is because there are important use cases that are enabled by20 ms.
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There is currently no API to determine audio latency over any path on an Android device atruntime. You can, however, use the following hardware feature flags to find out whether thedevice makes any guarantees for latency:
android.hardware.audio.low_latency
indicates a continuous output latency of 45 ms orless.android.hardware.audio.pro
indicates a continuous round-trip latency of 20 ms orless.
The criteria for reporting these flags is defined in the CDD in sections5.6 Audio Latencyand5.10 Professional Audio.
Here’s how to check for these features in Java:
Regarding the relationship of audio features, the
android.hardware.audio.low_latency
feature is a prerequisite for android.hardware.audio.pro
. A device can implementandroid.hardware.audio.low_latency
and not android.hardware.audio.pro
, but notvice-versa.Make no assumptions about audio performance
Beware of the following assumptions to help avoid latency issues:
- Don’t assume that the speakers and microphones used in mobile devices generally have goodacoustics. Due to their small size, the acoustics are generally poor so signal processing isadded to improve the sound quality. This signal processing introduces latency.
- Don't assume that your input and output callbacks are synchronized. For simultaneous inputand output, separate buffer queue completion handlers are used for each side. There is noguarantee 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 withproper buffer synchronization.
- Don't assume that the actual sample rate exactly matches the nominal sample rate. Forexample, if the nominal sample rate is 48,000 Hz, it is normal for the audio clock to advanceat a slightly different rate than the operating system
CLOCK_MONOTONIC
. This is becausethe audio and system clocks may derive from different crystals. - Don't assume that the actual playback sample rate exactly matches the actual capture samplerate, especially if the endpoints are on separate paths. For example, if you are capturing fromthe on-device microphone at 48,000 Hz nominal sample rate, and playing on USB audioat 48,000 Hz nominal sample rate, the actual sample rates are likely to be slightly differentfrom each other.
A consequence of potentially independent audio clocks is the need for asynchronous sample rateconversion. A simple (though not ideal for audio quality) technique for asynchronous sample rateconversion is to duplicate or drop samples as needed near a zero-crossing point. Moresophisticated conversions are possible.
Minimize input latency
This section provides suggestions to help you reduce audio input latency when recording with abuilt-in microphone or an external headset microphone.
- If your app is monitoring the input, suggest that your users use a headset(for example, by displaying a Best with headphones screen on first run). Notethat just using the headset doesn’t guarantee the lowest possible latency. You may need toperform other steps to remove any unwanted signal processing from the audio path, such as byusing the
VOICE_RECOGNITION
preset when recording. - Be prepared to handle nominal sample rates of 44,100 and 48,000 Hz as reported bygetProperty(String) forPROPERTY_OUTPUT_SAMPLE_RATE. Other sample rates are possible, but rare.
- Be prepared to handle the buffer size reported bygetProperty(String) forPROPERTY_OUTPUT_FRAMES_PER_BUFFER. Typical buffer sizes include 96, 128, 160, 192, 240, 256,or 512 frames, but other values are possible.
Minimize output latency
Use the optimal sample rate when you create your audio player
To obtain the lowest latency, you must supply audio data that matches the device's optimalsample rate and buffer size. For more information, seeDesign For Reduced Latency.
In Java, you can obtain the optimal sample rate from AudioManager as shown in the followingcode example:
Once you know the optimal sample rate, you can supply it when creating your player. This example uses OpenSL ES:
Note:
samplesPerSec
refers to the sample rate per channel in millihertz (1 Hz = 1000 mHz).Use the optimal buffer size to enqueue audio data
You can obtain the optimal buffer size in a similar way to the optimal sample rate, using theAudioManager API:
The
PROPERTY_OUTPUT_FRAMES_PER_BUFFER
property indicates the number of audio framesthat the HAL (Hardware Abstraction Layer) buffer can hold. You should construct your audiobuffers so that they contain an exact multiple of this number. If you use the correct numberof audio frames, your callbacks occur at regular intervals, which reduces jitter.It is important to use the API to determine buffer size rather than using a hardcoded value, because HAL buffer sizes differ across devices and across Android builds.
Don't add output interfaces that involve signal processing
Only these interfaces are supported by the fast mixer:
- SL_IID_ANDROIDSIMPLEBUFFERQUEUE
- SL_IID_VOLUME
- SL_IID_MUTESOLO
These interfaces are not allowed because they involve signal processing and will causeyour request for a fast-track to be rejected:
- SL_IID_BASSBOOST
- SL_IID_EFFECTSEND
- SL_IID_ENVIRONMENTALREVERB
- SL_IID_EQUALIZER
- SL_IID_PLAYBACKRATE
- SL_IID_PRESETREVERB
- SL_IID_VIRTUALIZER
- SL_IID_ANDROIDEFFECT
- SL_IID_ANDROIDEFFECTSEND
When you create your player, make sure you only add fast interfaces, as shown inthe following example:
Verify you're using a low-latency track
Enqueue 1 0 1 – Fast Light Weight Audio Player Portable
Complete these steps to verify that you have successfully obtained a low-latency track:
- Launch your app and then run the following command:
- Make a note of your app's process ID.
- Now, play some audio from your app. You have approximately three seconds to run thefollowing command from the terminal:
- Scan for your process ID. If you see an F in the Name column, it's on alow-latency track (the F stands for fast track).
Minimize warmup latency
When you enqueue audio data for the first time, it takes a small, but still significant,amount of time for the device audio circuit to warm up. To avoid this warmup latency, you canenqueue buffers of audio data containing silence, as shown in the following code example:
At the point when audio should be produced, you can switch to enqueuing buffers containingreal audio data.
Note: Constantly outputting audio incurs significant power consumption. Ensure that you stop theoutput in theonPause() method.Also consider pausing the silent output after some period of user inactivity.
Enqueue 1 0 1 – Fast Light Weight Audio Player Download
Additional sample code
To download a sample app showcasing audio latency, seeNDK Samples.