We get about milliseconds from the internal mic to the wired headset output. He was toying with an idea for an Android app and made his own research on it. Audio latency is what you first encounter while developing any audio application for Android using a microphone. That is one of those cases when the results of the preliminary research can either prove or disprove your concept before the development.
So, it was decided that our team makes our own audio latency analysis to check the client's calculations. Audio latency can include different factors like input latency, output latency, touchscreen lag and so on. We calculated only the latency of the device without any additional software processing, which is enough to see whether it is worth investigating the app idea further.
Round-trip audio latency is the time between capturing audio data, performing data zero processing in the application and playing the data. It depends on several factors:. Here, a special Android application performs recording and an immediate playback as fast as possible using the standard Java methods.
In order to calculate the latency we pair the Android device with the Bluetooth receiver, run this application on Android device, run an audio recording application on the PC Audacity with the stereo microphones connected, and just knock on the table near the microphone on the Android device.
As a result, the PC will record two knocks: the original one and yet another one that passed all processing path. Measuring the distance between two knocks in the audio editing software will give the total latency:. It is shown in the No Bluetooth column see the table below. The minimum value is ms, which is not so good.
Here is some statistics by Google which show that the best result for Nexus 9 with Android 5. It seems that a lot of Samsung devices support this technology.
Another thing that influences the total audio latency is Bluetooth latency. We get it by subtracting latency measured with the connected Bluetooth receiver and round-trip latency.
Such good values for Bluetooth latency are due to the fact that this device supports low-latency Bluetooth codec aptX. Here is the list of devices that support aptX. Note that there are no Nexus devices.
Here is the information about aptX latency. So in order to get total low latency, the device should be fast enough or be Samsung Professional Audio enabled and support aptX codec.
Note that Bluetooth receiver should support aptX too. We mentioned at the beginning that the data format also affects Bluetooth latency.Hetairos cost
Unfortunately, there is no simple way to force Android to use specific Bluetooth configuration. Here is the Issue Audio routing to the mono Bluetooth headset in the issues tracking system, which was changed to the Obsolete status on Mar 15,by the project manager without explanation.
So, currently, it is unknown whether Android supports mono headsets. What we can do is check what data format is used in the Bluetooth channel in the following way:. Enable the Bluetooth HCI snoop log option in the device Developer options note that it is available starting from Android 4.How to reset ford focus transmission
Install and launch the Wireshark application on your PC and open the copied file.To best understand the origins and fixes of Android audio latency, it is best to segregate contributions to total round-trip Android audio latency into two parts:. Latency is the lag or delay between between when an audio signal is sent, processed and received. Developers can should remember low latency best practices in Android app programming logic such as:. To help developers determine device latency, Superpowered maintains a free, open-source iOS and Android audio latency test app as well as continuously updated public database of latencies and native buffer sizes for Android devices.
All things being equal, the lower the buffer size, the lower the latency. However, all things are rarely equal in Android. Our latency test app measures the total round-trip latency of every component of the entire audio chain. NOTE : the Superpowered audio latency test app for Android and iOS does not use Superpowered, and in fact, does not do any audio processing whatsoever.
It measures the theoretical minimum round-trip latency of any device by measuring audio in and audio out. The latency data-table displays the best lowest measured value, if the dispersion is within 2x. We display a latency range if dispersion is outside of 2x. For full transparency, the latency test app's source is open and available on GitHub. We use the same code on all Android and iOS devices, and it works as follows:. The latency test app does not distinguish between input and output latency.
In theory, input latency and output latency are symmetrical, however devices may have an uneven split, due to different period sizes and period counts. Furthermore, many devices feature built-in mic processing to prevent echo, suppress noise and the Larsen effect feedback. Lastly, the latency test app only tests for one of the four main audio signal paths.
The four main possible audio signal path options, that can and do vary in latency on the same device, are:. A casual user might think each audio path returns a similar result, however, this is not the case.
Below we tested a few common devices to demonstrate the differences in latencies between audio paths.
OpenSL ES feature availability also varies wildly between the different devices. Wrappers are an abstraction layer above and around the audio features of an operating system and built on top of the existing audio environment. Wrappers attempt to make it easier to interact and program for audio APIs but have no impact on the fundamental technological capabilities such as latency, audio quality or power consumption.
Stack Overflow for Teams is a private, secure spot for you and your coworkers to find and share information. I'm guessing the difference stems from the well-known latency issues on Android. However, after reading around for a bit, I came upon this articlewhich states that:. Low-latency audio can be achieved using libpd, which is Pure Data library for Android.
Where can I find more information on the new low-latency audio in Jellybean?Low Latency Android Audio Programming Tips for Developers on Android and iOS
This is all I can find but it's sorely lacking in specific information. I'm using the AudioTrack API, and I'm not even sure if it should reap benefits from this improvement or if I should be looking into some other mechanism for audio playback. Should I look into using libpd? It seems to me like it's the only chance I have of achieving lower latencies, but since I've always thought of PD as an audio synthesis utility, is it really suited for a project that just grabs frames from a network stream and plays them back?
I'm not really doing any synthesizing. Am I following the wrong trail? As an additional note, before someone mentions OpenSL ES, this article makes it quite clear that no improvements in latency should be expected from using it :. In particular, use of OpenSL ES does not result in lower audio latency, higher scheduling priority, etc. For lowest latency on Android as of version 4. Use OpenSL.G pen gio not hitting
The Dalvik GC has a low amortized cost, but when it runs it takes more time than a low-latency audio thread can allow. Process audio in a buffer queue callback. The system runs buffer queue callbacks in a thread that has more favorable scheduling than normal user-mode threads. Make your buffer size a multiple of AudioManager. Otherwise your callback will occasionally get two calls per timeslice rather than one.
Unless your CPU usage is really light, this will probably end up glitching. Use the sample rate provided by AudioManager. Otherwise your buffers take a detour through the system resampler. Never make a syscall or lock a synchronization object inside the buffer callback.It is designed for high-performance audio applications that require low latency. Apps communicate with AAudio by reading and writing data to streams. AAudio moves audio data between your app and the audio inputs and outputs on your Android device.
Your app passes data in and out by reading from and writing to audio streamsrepresented by the structure AAudioStream. An audio device is a hardware interface or virtual endpoint that acts as a source or sink for a continuous stream of digital audio data. Don't confuse an audio device a built-in mic or bluetooth headset with the Android device the phone or watch that is running your app.
You can use the AudioManager method getDevices to discover the audio devices that are available on your Android device.
The method returns information about the type of each device. Each audio device has a unique ID on the Android device. You can use the ID to bind an audio stream to a specific audio device. However, in most cases you can let AAudio choose the default primary device rather than specifying one yourself. The audio device attached to a stream determines whether the stream is for input or output.
A stream can only move data in one direction. When you define a stream you also set its direction. When you open a stream Android checks to ensure that the audio device and stream direction agree. You can set the sharing mode explicitly when you create a stream. AAudio might perform sample conversion on its own. Conversion can happen in either direction. If your app processes audio input, it is wise to verify the input format and be prepared to convert data if necessary, as in this example:.
Set the audio stream configuration in the builder, using the builder functions that correspond to the stream parameters. These optional set functions are available:. Note that these methods do not report errors, such as an undefined constant or value out of range. If you do not specify the deviceId, the default is the primary output device.Latency is the time it takes for a signal to travel through a system. These are the common types of latency related to audio apps: Audio output latency is the time between an audio sample being generated by an app and the sample being played through the headphone jack or built-in speaker.
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 that touch 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 avoid warmup latency. It is difficult to measure audio input and output latency in isolation since it requires knowing exactly when the first sample is sent into the audio path although this can be done using a light testing circuit and an oscilloscope.
If you know the round-trip audio latency, you can use the rough rule of thumb: audio input and output latency is half the round-trip audio latency over paths without signal processing. Round-trip audio latency varies greatly depending on device model and Android build. You can get a rough idea of round-trip latency for Nexus devices by reading the published 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 this latency testing app.
This performs a round-trip latency test using the Larsen test. You can also view the source code for the latency testing app. Since the lowest latency is achieved over audio paths with minimal signal processing, you may also want to use an Audio Loopback Donglewhich allows the test to be run over the headset connector.
There is currently no API to determine audio latency over any path on an Android device at runtime. You can, however, use the following hardware feature flags to find out whether the device makes any guarantees for latency:. The criteria for reporting these flags is defined in the CDD in sections 5. Regarding the relationship of audio features, the android. A device can implement android. 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. This section provides suggestions to help you reduce audio input latency when recording with a built-in microphone or an external headset microphone. To obtain the lowest latency, you must supply audio data that matches the device's optimal sample rate and buffer size. For more information, see Design For Reduced Latency.
In Java, you can obtain the optimal sample rate from AudioManager as shown in the following code example:. Once you know the optimal sample rate, you can supply it when creating your player. You can obtain the optimal buffer size in a similar way to the optimal sample rate, using the AudioManager API:. You should construct your audio buffers so that they contain an exact multiple of this number.
If you use the correct number of 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. These interfaces are not allowed because they involve signal processing and will cause your request for a fast-track to be rejected:. When you create your player, make sure you only add fast interfaces, as shown in the following example:.
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 can enqueue 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 containing real audio data. Note: Constantly outputting audio incurs significant power consumption.
Audio Latency for App Developers
Ensure that you stop the output in the onPause method.Account Options Sign in. Top charts. New releases. Add to Wishlist. SoundWire lets you stream any music or audio "what you hear now" from your Windows or Linux PC to your Android mobile devices.
Use it as: - A remote speaker or wireless headphones - A way to listen to music and movies from your computer anywhere around your house, or further over cell networks - A wireless extension of live audio from your PC-based music system SoundWire does audio mirroring audio cast. There are other uses too SoundWire can work as a baby monitor or listening device with a computer such as a netbook that has a built-in microphone.Brother ppd file chromebook
Raspberry Pi is also supported. Features - Live audio capture and streaming - Excellent sound quality The most common reason for connection problems is incorrect firewall settings at the PC or router. The free version of the application identifies itself by voice every 45 minutes and displays ads. There is a 10 minute compression trial in the free version.
The full version enables unlimited Opus audio compression, can handle multiple clients, and has no ads or voice identification.
It also has a special Pro Mode to set and display buffer latency precisely in milliseconds. Please consider purchasing the full version of SoundWire if you'd like to support the developer. After that it will update normally through Google Play. May be disabled in settings. Reviews Review Policy. Thanks for using SoundWire!Maixduino examples
New setting "disable fast Android audio path" should be selected to prevent low-latency audio from disabling equalizer. View details. Flag as inappropriate.This is a rather long reference, and not all of it will be relevant to you; but you will need to consult it for details on the API.
Open this link in a browser:. You'll want to skim the whole document, but pay special attention to the "Performance" subsection of the "Programming notes" section.Hj�lp til din regning
The site source. However, there is a wealth of useful information about latency at this site, so you may want to review it. See the articles at Audio Latency. If you have questions about how to use Android native audio, you can ask at the discussion group android-ndk. Content and code samples on this page are subject to the licenses described in the Content License. Google is committed to advancing racial equity for Black communities.
See how. Read the supporting documentation. Check for API level 9 or higher. Check for feature android. Use the recommended native buffer size and sample rate returned by android. String Note : the same buffer size and sample rate should also be used for input. Keep your callback handlers short, without bursty CPU usage or unbounded blocking. Avoid priority inversion. Consider using non-blocking algorithms to communicate between input and output callback handlers, and between the callback handlers and the rest of your application.
Figure 1. Venn diagram Other resources source.
Audio latency: buffer sizes Days of Google Dev Describes the relationship between audio latency, buffer sizes, and task scheduling.
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