I want to use AudioRecord and AudioTrack classes(in SDK 1.5) in my program. Where can I find how to use it. Is there any API demo program for this?. If not it is greatly appreciated if someone can post a sample code in this forum.
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My application uses AudioTrack in streaming mode and AudioRecord simultaneously.My problem is that I start them at the same time, but I have no guarantee that they will actually start playback ecording in the same timestamp. The reason I need this kind of accuracy is because I use echo cancellation (subtracting audio played to speaker from the recording).My echo canceller doesn't require an exact delay, but the delay introduced between AudioTrack and AudioRecord moves in the range of 250ms (between different runs), and that's too much - each run is different because they start themselves asynchronously. The API doesn't provide me a way to make sure they start in the same time, so I thought about measuring this starting delay somehow and then using it in my calculation.Does anyone have an idea how to do that, utilizing their API or in any other way?
View 2 Replies View RelatedI currently use the AudioTrack and AudioRecord classes in Android.I use the pure PCM data but I was wondering what my options are for other codecs?From this page it seems I can only encode and decode using AMR narrowband?The code runs properly but I'm wondering does it actually encode the Audio as AMR_NB and if this is not a proper way to do it?I was getting a buffer overflow when using raw PCM but none have appeared since using the new code with the MediaRecorder.AudioEncoder.AMR_NB used instead of the AudioFormat.PCM
View 2 Replies View RelatedI need to stop the AudioTrack, regardless what's in the buffer, immediately when executing the stop/release. The AudioTrack plays in the STREAM mode. It appears that the AudioTrack finishes playing what's in the buffer and than stops. It seems that it's such a fundamental flaw since there are so many audio applications out there.
View 4 Replies View RelatedI'm trying to figure out how to use the AudioRecord class.I created a callback with a logging message, but don't ever see it called.Do you see anything wrong with what I'm doing? Do you have an example of how to use the API?
View 6 Replies View RelatedI'm currently initiating the AudioTrack() for MODE_STREAM. I load some bytes using the write() and I see that it is working but once hit the play() call nothing happens. what is the matter? should I do something else that is not obvious? Any tips Google friends? My settings: streamType - STREAM_MUSIC sampleRateInHz - 44100 channel Config - CHANNEL _CONFIGURATION_STEREO audioFormat - ENCODING_PCM_16BIT bufferSizeInBytes - 256K the total size (in bytes) of the buffer where audio data is read from for playback. mode - MODE_STREAM.
View 21 Replies View RelatedI currently have a Loop back program for testing Audio on Android devices.It uses AudioRecord and AudioTrack to record PCM audio from the Mic and play PCM audio out the earpiece. So as you can see in the creation of the AudioTrack and AudioRecord the Encoding is supplied via the AudioFormat but this only allows 16 bit or 8 bit PCM.I have my own G711 Codec implementation now and I want to be able to encode the audio from the Mic and decode it going into the EarPiece, So I have encode(short lin[], int offset, byte enc[], int frames) and decode(byte enc[], short lin[], int frames) methods but I'm unsure as to how to use them to encode and the decode the audio from the AudioRecord and AudioTrack.
View 2 Replies View RelatedI used AudioRecord on emulator and phone. I found that AudioRecord is not working on phone: 1> the min-buffer-size on simulator is much smaller than phone 2> The audio data I read from phone is all zero.
View 2 Replies View RelatedI'm doing a voice recording using AudioRecord class and the main requirement that the frequency must be 16000 Hz. I'm using only emulator in development process for now and met the problem that I can't set frequency more than 8000, otherwise I get bufferSize -2, and can't create instance of AudioRecord with such buffer. Thank you to Szabolcs Vrbos, he told me that this problem is only with emulator and several Samsung phones, device can record with higher frequency.I need a word from Android engineers that I will be able to record voice with 16000 Hz frequency on for instance HTC Hero (Android 1.5)or similar device with os 1.5 or higher.
View 2 Replies View Relatedthe location where the audioflinger, surfaceflinger and cameraservice talk to MediaRecorder's APIs? Is there any relation between MediaRecorder and AudioRecord?
View 6 Replies View RelatedWe have been getting reports of audio recording not working on Samsung Moment phones on 2.1-update1.
int bufferSize = AudioRecord.getMinBufferSize(sampleRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT); this.audioRecord = new AudioRecord(MediaRecorder.AudioSource.MIC, sampleRate, AudioFormat.CHANNEL_IN_MONO, AudioFormat.ENCODING_PCM_16BIT, bufferSize);
generates these logs: I/AudioPolicyManager( 1889): getInput() inputSource 1, samplingRate 16000, format 1, channels 10, acoustics 0 W/AudioHardwareALSA( 1889): openInputStream : mInput already exists!! E/AudioRecord( 2618): Could not get audio input for record source 1 E/AudioRecord-JNI( 2618): Error creating AudioRecord instance: initialization check failed. E/AudioRecord-Java( 2618): [ android.media.AudioRecord ] Error code -20 when initializing native AudioRecord object. I can't find the "mInput already exists". anywhere in the source base, could this perhaps be an error message included by the OEM?
I want to play music from online mp3 link.I am reading the music data into a stream and trying to play it using audio track. But itz giving only noise.No music i could listen.Same buffer I can play in Media Player,But i could not play in audio track.
View 16 Replies View RelatedI am getting this clicking sound when playing static pcm data. Anybody out there is having the same problem? This is not the data issue.
View 7 Replies View RelatedI'm in the process of wring a music player that can play SID tracks (music from the Commodore 64), and everything is progressing nicely, except that I sometimes get audio stuttering.The stuttering only happens when the player is in the background, and another application has focus. It's most noticeable when a cpu heavy app(like Google Earth) is running in the foreground.I have tried adjusting the priority of my thread and increasing the AudioTrack buffer size, but nothing has solved the stuttering problem.I believe the problem is that generating the audio takes quite a lot of cycles (I'm at 15-30% cpu load), and that my audio thread is being starved and cant keep up with AudioTrack's constant need for data.I'm must be doing something wrong, I can't imagine that gui threads should be able to starve a high priority audio thread.I hope someone have some ideas on how to get my player stutter free, as I'm at a loss :)
View 9 Replies View RelatedIf I create and release multiple AudioTrack objects, the GREF count continually increases which eventually causes an application crash. Even with a forced garbage collection, I end up with the same high GREF count at the end of the loop.In my application, I continuously create AudioTracks from variable lenght buffers.I guess I can try and use a fixed size pool of AudioTracks each with a buffer large enough to fit my longest sound, and try and reuse them.Is there a better/correct way to completely clear resources used by an AudioTrack?
View 5 Replies View RelatedThere is a AudioTrack Class in SDK 1.5 which can play PCM buffer directly.But it is not exist in SDK 1.1. I want to play PCM buffer in SDK1.1,but i don't know how.
View 2 Replies View RelatedI am trying to determine if I should go the ndk way to work on a music synth app, but first wanted to do a sanity check in the java layer by testing how many sine tones i could play concurrently, and so I tried this. And it's pretty low: I can only play about 4 or 5 tones before the obtainBuffer timeouts start increasing in number. Also, changing the media volume while the tones are playing disrupts the output.
View 3 Replies View RelatedI am trying to resolve the clicking problem while using the AudioTrack. It takes place only when playing the first part of the wave (pcm) buffer when the offset to the write is 0 (zero). track.write(buffer, offset, minBufferSize); It tells me that there is something wrong with the pcm data. Could it be because the wave data, I generate using Audacity, has some header as opposed to the pure pcm which does not? I was looking for the converter tools to generate pure pcm but found none. Applications I found generate only wave data so I am not able to test this assumption. At this point I am sure that filling the buffer is correct. Any ideas?
View 2 Replies View RelatedI'm currently writing an app that uses a Service. A thread in the service generates an infinite stream of audio, and writes it to the AudioTrack. I want the audio to keep playing indefinitely until the user stops it. Based on my understanding of the PowerManager class, it looks like I should need a PARTIAL_WAKE_LOCK to keep the CPU awake so the audio can continue playing.
However, I left the audio playing on my phone with no wake lock, running on battery, in airplane mode, with all five items on the power control widget disabled, and the audio just kept right on playing for at least 15 minutes. Afterwards, I looked at the battery history, and it showed "Running (100%)", even though the Partial wake usage was negligible. Is there something special about AudioTrack which prevents the CPU from entering its deep sleep state? Should it be necessary for my service to grab a PARTIAL_WAKE_LOCK and its associated permission?
It doesn't seems as if the the stop and release immediately terminates the audio played by the MODE_STREAM track. It looks like it finishes playing the rest of the buffer and then stops.
View 3 Replies View RelatedHow can I get the amplitude/volume when recording with AudioRecord?
View 2 Replies View RelatedI've been having an issue with using AudioRecord for Android. I've read as much as I can find online about it, but I cannot seem to get a good initialization. I have tried the Android 2.2 emulator, 1.5 emulator and my phone, an HTC Incredible running Froyo. The emulators and my phone fail initialization. I've tried sampling rates of 8000, 11025, and 44100, formats of CHANNEL_IN_MONO/STEREO and CHANNEL_CONFIGURATION_MONO/STEREO, 8bit and 16bit encoding (8 bit makes the getMinBufferSize fail), and Audio Source of MIC and DEFAULT. All result in the variable test becoming 0 after running a get state(failed initialization). It seems from everything I've read that this should correctly initialize the object. I have played around with the multiplier on buflen, to have it range from 512 (the result of the function) to 102400 because I had heard that HTC devices require something above 8192. For testing my problem I made a new, small project that recreates my problem as simply as possible. I pull out the constants needed into local ints then run the constructor and access the getState method for checking if it worked. Code...
View 1 Replies View RelatedHere is my AudioRecorder class, using audio record, why is it not producing any sound data?
import java.io.FileOutputStream; import java.io.IOException; import android.content.Context; import android.media.AudioFormat; import android.media.AudioRecord; import android.media.MediaRecorder; public class AudioRecorder implements Runnable {public boolean isRecording = false; byte[] tempBuffer = new byte[AudioRecord.getMinBufferSize(44100, AudioFormat.CHANNEL_CONFIGURATION_STEREO,......
I am using the AudioRecord class to analize raw pcm bytes as it comes in the mic.
So thats working nicely. Now i need convert the pcm bytes into decibel.
I have a formula that takes sound presure in Pa into db. db = 20 * log10(Pa/ref Pa)
So the question is the bytes i am getting from audiorecorder from the buffer what is it is it amplitude pascal sound pressure or what.
I tried to putting the value into te formula but it comes back with very hight db so i do not think its right
I am doing recording in following way and storing byte array in a binary file. But the binary file I am getting is not in any format. what can be the problem? code...
View 3 Replies View RelatedMy Android Java Application needs to record audio data into the RAM and process it. This is why I use the class "AudioRecord" and not the "MediaRecorder" (records only to file).
Till now, I used a busy loop polling with "read()" for the audio data. this has been working so far, but it peggs the CPU too much. Between two polls, I put the thread to sleep to avoid 100% CPU usage. However, this is not really a clean solution, since the time of the sleep is not guaranteed and you must subtract a security time in order not to loose audio snippets. This is not CPU optimal. I need as many free CPU cycles as possible for a parallel running thread.
Now I implemented the recording using the "OnRecordPositionUpdateListener". This looks very promising and the right way to do it according the SDK Docs. Everything seems to work (opening the audio device, read()ing the data etc.) but the Listner is never called.
I am working with a real Device, not under the Emulator. The Recording using a Busy Loop basically works (however not satifiying). Only the Callback Listener is never called.
Here is a snippet from my Sourcecode:
CODE:.........................
On the nexus one, my app goes into Audio Record Stanby randomly after 0-40 seconds. I'm using the AudioRecord class, and while the standby is reported in the terminal, the AudioRecord instance reports the same states of RECORDSTATE_RECORDING and STATE_INITIALIZED throughout the entire sequence.
The bottom line is that when this anomaly occurs, audio is not captured. and this is what im working to prevent was only able to reproduce on nexus one android2.1 not g1 android1.6
I have looked extensively through the Android C/C++ libraries such as at this file: http://android.git.kernel.org/?p=platform/hardware/msm7k.git;a=blob_p... and [url]
Most interesting to me, is this line in AudioFlinger: [url]
Specifically the bool AudioFlinger::RecordThread::threadLoop()
As you can see, minput->standby() occurs in 4 different places in the code.
I've seen posts regarding problems with the AudioRecord OnRecordPositionUpdateListener, but I haven't seen any real answers.
I'm using AudioRecord to record from the mic, and I want to get the input from the mic periodically so I can check sound levels and update a meter.
I set up the listener like this:
CODE:...........
What happens is that onMarkerReached gets called once, after the recorder is released, so that's not very useful.
Do I need to change the marker position and notification period? Is the listener working? Is there another way to accomplish periodically getting the mic input?
I am getting the 'AudioTrack::start called from thread' when starting an AudioTrack. I am using it in a thread. I asume I shouldn't but why? I see it as a major limitation.
View 4 Replies View RelatedIs the AudioTrack fundamentally designed to be a singleton? Methods on the OnPlaybackPositionUpdateListener receive just one argument, which is the AudioTrack itself. And since the AudioTrack does not have getTag/setTag methods I don't see any way to distinguish among multiple AudioTracks in the OnPlaybackPositionUpdateListener methods short of keeping track of references. So, if it's so why it's not implemented as a singleton? Any thoughts. I am trying to implement concurrent use of the AudioTrack. Anybody has any experience with this approach? Am I wasting time?
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