Getminbuffersize

Works Great But I want to give delay in replying the sound Plz Help me in this Thanks in advance. Some one help me plz!! This was an nice and amazing and the given contents were very useful and the precision given here is good. Thanks a lot very much for the high quality and results-oriented help. Fantastic work! This is the type of information that should follow collective approximately the web. Embarrassment captivating position Google for not positioning this transmit higher!

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getminbuffersize

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Sign up. Branch: master. Find file Copy path. Raw Blame History. Manifest ; import android.

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PackageManager ; import android. AudioAttributes ; import android. AudioFormat ; import android. AudioManager ; import android. AudioRecord ; import android. AudioTrack ; import android. MediaRecorder ; import android. AsyncTask ; import android. Build ; import android. Bundle ; import android. Environment ; import android. NonNull ; import android. ActivityCompat ; import android. ContextCompat ; import android. AppCompatActivity ; import android. Log ; import android. View ; import android.

Button ; import com.I mentioned the class AudioTrack a couple of times already. AudioTrack is available since Android 1.

You can use it in two modes: static and streaming. I will only look at the streaming mode here. Streaming mode means that you permanentley write new PCM data to the hardware, the framework will queue it in a buffer and play it back for you.

An AudioTrack instance can either be in mono or stereo mode. Pretty simple eh? The first thing we do is to get the minimum buffer size for the AudioTrack instance we are going to create. This is achieved by a call to AudioTrack. If the buffer is full it is flushed to the audio hardware. Now, in the next line we instantiate an AudioTrack. The first parameter dictates which audio stream our samples are going to be written to. For the other stream types refer to the documentation.

The next three parameters say what sampling rate we want to have, wheter we want the track to be mono or stereo and which PCM encoding we want to use. The last parameter says wheter this AudioTrack is static or a streaming one, we want it to be a streaming one. All the code in the onset detection tutorial worked with PCM data encoded as floats in the range [-1,1]. We want to emulate this here so i wrote a little method called AndroidAudioDevice.

For this the float samples have to be converted to bit signed PCM which is done in the AudioDevice. Upon the next write it will start playback again introducing a wake up lag. The lag will be ms in that case.

Synthesizing sounds each time the screen is touched for example is a bad idea as the lag is more than noticeable. Another property of AudioTrack is that the AudioTrack. If you want to use it in a game you should do all your audio mixing in a seperate thread. Still, the class is pretty niffty and it makes it easy to port all the examples from the onset detection tutorial to Android.

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I tested it with the WaveDecoder class and it worked like a charm, not eating up to much system resources while doing its thing. You can also try to use the decoders included in the tutorial framework, however, the pure Java MP3 decoder will be to slow. Only the WaveDecoder works acceptably.

AudioRecord

It seems that the clicking sound issue had not been resolved. I think it is a physics problem. The clicking sound is a good indicator that the start or stop of the wave file did not start or end at an angle of zero.

It will be fairly easy to control with a single tone, but if you have a wave file with more than one tone combined, the only way to stop or start this without a click is to control the amplitude of the sound during startup and ending.

Sorry I am not a good programmer and I cannot give you a lot of code support, but I remember reading about a variable used to control the amplitude of audio output in the SDK. Looks like I have given credit to Mike instead of you for posting the original code.

My apology to you. I have a mono audio file khz and 16bit PCM. There is none. Problem is it still has the same latency characteristics as Audiotrack. It may or may not help to not call track. I know this seems to be a common problem. This is either missing samples or a hardware problem?This post is a continuation of the post related to the implementation of wave generation functionality in the PSLab Android App.

In this post, the subject matter of discussion is the way to fill the audio buffer so that the resulting wave generated is either a Sine Wavea Square Wave or a Sawtooth Wave. The waves we are trying to generate are periodic waves. Periodic Wave: A wave whose displacement has a periodic variation with respect to time or distance, or both. Thus, the problem reduces to generating a pulse which will constitute a single time period of the wave.

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Suppose we want to generate a sine wave; if we generate a continuous stream of pulses as illustrated in the image below, we would get a continuous sine wave. This is the main concept that we shall try to implement using code.

AudioTrack object is initialised using the following parameters:. Depending on the values in the audio buffer, the wave is generated by the AudioTrack object. Therefore, to generate a specific kind of wave, we need to fill the audio buffer with some specific values. The values are governed by the wave equation of the signal that we want to generate.

Repeating this pulse continuously, we will get a square wave.

getminbuffersize

Ramp signals increases linearly with time. A Ramp pulse has been illustrated in the image below:. We need repeated ramp pulses to generate a continuous sawtooth wave.

getminbuffersize

Finally, when the audio buffer is generated, write it to the audio sink for playback using write method exposed by the AudioTrack object. Another application which can be implemented by hacking the audio jack is Wave Generation. We can generate different types of signals on the wires connected to the audio jack using the Android APIs that control the Audio Hardware. In this post, I will discuss about how we can generate wave by using the Android APIs for controlling the audio hardware.

Simply cut open the wire of a cheap pair of earphones to gain control of its terminals and attach alligator pins by soldering or any other hack jugaad that you can think of. After you are done with the tinkering of the earphone jack, it should look something like shown in the image below.Jump to navigation. As one of the interfaces on mobile devices and tablets, the key function of the audio jack is to play music.

However, its other usage cannot be ignored—the audio jack can also be used to transmit data. More usages of the audio jack to connect devices are being developed all the time. Since wearable devices and peripherals have broad market prospects, I believe that the audio jack will be the prime data communication portal. In this article I will explain more details of this new feature.

Figure 1. When we send a 0x00FF data value, the first step is to convert the digital data value to an analog signal. We need to modulate the data value. Normally, we use a sine wave carrier for the analog signal.

Figure 2. FSK [5] modulation signal [6]. The following code sends data to the buffer using the audioTrack function. As a receiver, we need to translate the analog signal to a data value, demodulate the signal to remove the carrier signal, and decode the data by protocol.

The protocol can be a public data format or private definition protocol. Figure 3. Demodulate the signal 6]. Obviously, power is required to drive the circuit for audio jack peripherals. For example, the L-channel sends data information. The R-channel sends sustained square or sine waveforms. Androlirc [9] is a Github-based project. Its function is to use the audio jack interface to send infrared commands. We can study this project to understand audio jack data communication.

Androlirc uses the LIRC [10] library to create a data buffer. Androlirc can use the LIRC library as a data encapsulation or data capsulation. Across the marketplace, you can find many infrared coding types, such as the RC-5 and RC-6 protocols. In the example, we use the RC-5 protocol to control a TV set. First, we need to modulate the data value using a 38k frequency sine waveform to generate buffer data; then we use the Android audio API function to play the audio buffer data.

Meanwhile we can use one of two channels to play a sine or square waveform to power on the peripherals hardware. Hijack [12] is a University of Michigan student project. The Hijack platform enables a new class of small, cheap, phone-centric sensor peripherals that support plug-and-play operations.

Figure 4 shows a smartphone displaying the indoor temperature from the audio jack-based peripheral temperature sensor.

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Figure 4. Temperature value from Quick-Jack. Figure 5. Wearables and smart device peripherals are becoming more prevalent in the consumer market. The audio jack as a data communication function is being adopted by more and more ODMs and iMakers.

Perhaps in the future, audio jack data communication functions will be universally supported by smartphone OSs.The AudioRecord class manages the audio resources for Java applications to record audio from the audio input hardware of the platform. This is achieved by "pulling" reading the data from the AudioRecord object. The application is responsible for polling the AudioRecord object in time using one of the following three methods: read byte[], int, intread short[], int, int or read ByteBuffer, int.

The choice of which method to use will be based on the audio data storage format that is the most convenient for the user of AudioRecord. Upon creation, an AudioRecord object initializes its associated audio buffer that it will fill with the new audio data.

Record and Play the sound simultaneously (live stream over Bluetooth speaker) in your Android App?

The size of this buffer, specified during the construction, determines how long an AudioRecord can record before "over-running" data that has not been read yet. Data should be read from the audio hardware in chunks of sizes inferior to the total recording buffer size. Returns the configured audio data format. Returns the configured channel configuration. Returns the minimum buffer size required for the successful creation of an AudioRecord object.

Note that this size doesn't guarantee a smooth recording under load, and higher values should be chosen according to the expected frequency at which the AudioRecord instance will be polled for new data. Returns the state of the AudioRecord instance. This is useful after the AudioRecord instance has been created to check if it was initialized properly. This ensures that the appropriate hardware resources have been acquired. Reads audio data from the audio hardware for recording into a direct buffer.

If this buffer is not a direct buffer, this method will always return 0. Releases the native AudioRecord resources. The object can no longer be used and the reference should be set to null after a call to release.

Using the Audio Jack as Data Interface on Android* Systems

Sets the listener the AudioRecord notifies when a previously set marker is reached or for each periodic record head position update. Use this method to receive AudioRecord events in the Handler associated with another thread than the one in which you created the AudioTrack instance.

getminbuffersize

Starts recording from the AudioRecord instance when the specified synchronization event occurs on the specified audio session. Invoked when the garbage collector has detected that this instance is no longer reachable. The default implementation does nothing, but this method can be overridden to free resources. Note that objects that override finalize are significantly more expensive than objects that don't. Finalizers may be run a long time after the object is no longer reachable, depending on memory pressure, so it's a bad idea to rely on them for cleanup.

Note also that finalizers are run on a single VM-wide finalizer thread, so doing blocking work in a finalizer is a bad idea.

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A finalizer is usually only necessary for a class that has a native peer and needs to call a native method to destroy that peer. Even then, it's better to provide an explicit close method and implement Closeableand insist that callers manually dispose of instances.

This works well for something like files, but less well for something like a BigInteger where typical calling code would have to deal with lots of temporaries.By using our site, you acknowledge that you have read and understand our Cookie PolicyPrivacy Policyand our Terms of Service.

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I have tried setting sampleRate to, and I have also tried changing AudioFormat. I have also tried 16 bit encoding instead of 8 bit. From the platform source file AudioRecord. Learn more. Why does AudioRecord. Ask Question. Asked 9 years, 2 months ago.

Active 8 years, 2 months ago. Viewed 10k times. I am testing this on a Samsung Galaxy S i The native sample rate is Hz, as returned by AudioTrack. I keep getting -2 as a result. Why is this happening? Octavian A. Damiean Tom Tom 8, 26 26 gold badges silver badges bronze badges. Hi, did this ever get resolved, or any insights?

Thank you. Active Oldest Votes. Reuben Scratton Reuben Scratton 37k 8 8 gold badges 70 70 silver badges 84 84 bronze badges. Thanks for the info.

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