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• Android sound starter project for Eclipse

  - by Jerry

  Is there an example Hello Sound project for Android that will show how to create beep sounds of a variable tone and duration?

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• How to produce precisely-timed tone and silence?

  - by Bob Denny

  I have a C# project that plays Morse code for RSS feeds. I write it using Managed DirectX, only to discover that Managed DirectX is old and deprecated. The task I have is to play pure sine wave bursts interspersed with silence periods (the code) which are precisely timed as to their duration. I need to be able to call a function which plays a pure tone for so many milliseconds, then Thread.Sleep() then play another, etc. At its fastest, the tones and spaces can be as short as 40ms. It's working quite well in Managed DirectX. To get the precisely timed tone I create 1 sec. of sine wave into a secondary buffer, then to play a tone of a certain duration I seek forward to within x milliseconds of the end of the buffer then play. I've tried System.Media.SoundPlayer. It's a loser because you have to Play(), Sleep(), then Stop() for arbitrary tone lengths. The result is a tone that is too long, variable by CPU load. It takes an indeterminate amount of time to actually stop the tone. I then embarked on a lengthy attempt to use NAudio 1.3. I ended up with a memory resident stream providing the tone data, and again seeking forward leaving the desired length of tone remaining in the stream, then playing. This worked OK on the DirectSoundOut class for a while (see below) but the WaveOut class quickly dies with an internal assert saying that buffers are still on the queue despite PlayerStopped = true. This is odd since I play to the end then put a wait of the same duration between the end of the tone and the start of the next. You'd think that 80ms after starting Play of a 40 ms tone that it wouldn't have buffers on the queue. DirectSoundOut works well for a while, but its problem is that for every tone burst Play() it spins off a separate thread. Eventually (5 min or so) it just stops working. You can see thread after thread after thread exiting in the Output window while running the project in VS2008 IDE. I don't create new objects during playing, I just Seek() the tone stream then call Play() over and over, so I don't think it's a problem with orphaned buffers/whatever piling up till it's choked. I'm out of patience on this one, so I'm asking in the hopes that someone here has faced a similar requirement and can steer me in a direction with a likely solution.

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• Playing an arbitrary tone with Android.

  - by fiXedd

  Is there any way to make Android emit a sound of arbitrary frequency (meaning, I don't want to have pre-recorded sound files)? I've looked around and ToneGenerator was the only thing I was able to find that was even close, but it seems to only be capable of outputting the standard DTMF tones. Any ideas?

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• No mic activity with setLoopBack set to false - AS3

  - by Franky

  Trying to figure out why setloopback needs to be set to true for microphone activity to be detected. The problem is the echo feedback when using a macbook with a built in mic. If anyone has some ideas about this let me know. Right now I'm experimenting with toggling gain, depending on activity to simulate echo reduction. Not optimal though. @lessfame

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• What is the best way to merge mp3 files?

  - by Dan Williams

  I've got many, many mp3 files that I would like to merge into a single file. I've used the command line method copy /b 1.mp3+2.mp3 3.mp3 but it's a pain when there's a lot of them and their namings are inconsistent. The time never seems to come out right either.

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• Piping SoX in Python - subprocess alternative?

  - by Cochise Ruhulessin

  I use SoX in an application. The application uses it to apply various operations on audiofiles, such as trimming. This works fine: from subprocess import Popen, PIPE kwargs = {'stdin': PIPE, 'stdout': PIPE, 'stderr': PIPE} pipe = Popen(['sox','-t','mp3','-', 'test.mp3','trim','0','15'], **kwargs) output, errors = pipe.communicate(input=open('test.mp3','rb').read()) if errors: raise RuntimeError(errors) This will cause problems on large files hower, since read() loads the complete file to memory; which is slow and may cause the pipes' buffer to overflow. A workaround exists: from subprocess import Popen, PIPE import tempfile import uuid import shutil import os kwargs = {'stdin': PIPE, 'stdout': PIPE, 'stderr': PIPE} tmp = os.path.join(tempfile.gettempdir(), uuid.uuid1().hex + '.mp3') pipe = Popen(['sox','test.mp3', tmp,'trim','0','15'], **kwargs) output, errors = pipe.communicate() if errors: raise RuntimeError(errors) shutil.copy2(tmp, 'test.mp3') os.remove(tmp) So the question stands as follows: Are there any alternatives to this approach, aside from writing a Python extension to the Sox C API?

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• Pitch detection and change java

  - by omegas27

  Hello, I'm french so I'm sorry if you have trouble to understand some of my sentences. Aniways, I saw in some topics that the pitch could be fetected thanks to the Fourier transform but I didn't really understand how to implement it. Moreover, I didn't find how to change the pitch of a wav file and if possibl ,a mp3 file I am listening to music using javaSound for the wav and JLayer for the mp3. Thanks

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• Extracting note onset from MIDI

  - by Dolphin

  Hi I need to extract musical features (note details-pitch, duration, rhythm, loudness, note start time) from a polyphonic (having 2 scores for treble and bass - bass may also have chords) MIDI file. I'm using the jMusic API to extract these details from a MIDI file. My approach is to go through each score, into parts, then phrases and finally notes and extract the details. With my approach, it's reading all the treble notes first and then the bass notes - but chords are not captured (i.e. only a single note of the chord is taken), and I cannot identify from which point onwards are the bass notes. So what I tried was to get the note onsets (i.e. the start time of note being played) - since the starting time of both the treble and bass notes at the start of the piece should be same - But I cannot extract the note onset using jMusic API. Each time it shows 0.0. Is there any way I can identify the voice (treble or bass) of a note? And also all the notes of a chord? How is the voice or note onset for each note stored in MIDI? Is this different for each MIDI file? Any insight is greatly appreciated. Thanks in advance

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• Playing Multiple sounds at the same time in Android

  - by Wrapper

  I am unable to use the following to code to play multiple sounds/beeps simultaneously. In my onclicklistener I have added ... public void onClick(View v) { mSoundManager.playSound(1); mSoundManager.playSound(2); } ... But this plays only one sound at a time, sound with index 1 followed by sound with index 2. How can I play atleast 2 sounds simultaneously using this code whenever there is an onClick() event? public class SoundManager { private SoundPool mSoundPool; private HashMap<Integer, Integer> mSoundPoolMap; private AudioManager mAudioManager; private Context mContext; public SoundManager() { } public void initSounds(Context theContext) { mContext = theContext; mSoundPool = new SoundPool(4, AudioManager.STREAM_MUSIC, 0); mSoundPoolMap = new HashMap<Integer, Integer>(); mAudioManager = (AudioManager)mContext.getSystemService(Context.AUDIO_SERVICE); } public void addSound(int Index,int SoundID) { mSoundPoolMap.put(1, mSoundPool.load(mContext, SoundID, 1)); } public void playSound(int index) { int streamVolume = mAudioManager.getStreamVolume(AudioManager.STREAM_MUSIC); mSoundPool.play(mSoundPoolMap.get(index), streamVolume, streamVolume, 1, 0, 1f); } public void playLoopedSound(int index) { int streamVolume = mAudioManager.getStreamVolume(AudioManager.STREAM_MUSIC); mSoundPool.play(mSoundPoolMap.get(index), streamVolume, streamVolume, 1, -1, 1f); } }

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• Android PCM Bytes

  - by Pintac

  Hi 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 thanks

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• AVAudioPlayer currentTime problem

  - by StrAbZ

  Hi, I'm trying to use the audioplayer with a slider in order to seek into a track (nothing complicated). But I have a weird behavior... for some value of currentTime (between 0 and trackDuration), the player stop playing the track, and goes into "audioPlayerDidFinishPlaying:successfully:" with successfully to NO. And it did not go into "audioPlayerDecodeErrorDidOccur:error:" It's like it can't read the time i'm giving to it. For exemple the duration of the track is: 295.784424 seconds i set the currentTime to 55.0s (ie: 54.963878 or 54.963900 or 54.987755, etc... when printed as %f). The "crashes" always happen when the currentTime is 54.987755... and I really don't understand why... So if you have any idea... ^^

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• SoundPlayer causing Memory Leaks?

  - by Nick Udell

  I'm writing a basic writing app in C# and I wanted to have the program make typewriter sounds as you typed. I've hooked the KeyPress event on my RichTextBox to a function that uses a SoundPlayer to play a short wav file every time a key is pressed, however I've noticed after a while my computer slows to a crawl and checking my processes, audiodlg.exe was using 5 GIGABYTES of RAM. The code I'm using is as follows: I initialise the SoundPlayer as a global variable on program start with SoundPlayer sp = new SoundPlayer("typewriter.wav") Then on the KeyPress event I simply call sp.Play(); Does anybody know what's causing the heavy memory usage? The file is less than a second long, so it shouldn't be clogging the thing up too much.

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• Getting following exception javax.sound.sampled.LineUnavailableException: line with format ULAW 800

  - by angelina

  Dear All, I tried to play and get duration of a wave file using code below but got following exception.please resolve.I m using a wave file format. URL url = new URL("foo.wav"); Clip clip = AudioSystem.getClip(); AudioInputStream ais = AudioSystem.getAudioInputStream(url); clip.open(ais); System.out.println(clip.getMicrosecondLength()); **javax.sound.sampled.LineUnavailableException: line with format ULAW 8000.0 Hz, 8 bit, mono, 1 bytes/frame, not supported.**

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• How can I get latency info from Android's AudioTrack class?

  - by Ryan

  I've noticed that the C++ classes underlying the AudioTrack and AudioRecord APIs in Android both have a latency() method that is not exposed via JNI. As far as I can see, the latency() method in AudioRecord still does not take into account the hardware latency (they have a TODO comment for that), but the latency() method in AudioTrack does add in the hardware latency. I absolutely need to get this latency value from AudioTrack. Is there any possible way I can do this? I don't care what kind of crazy hack is needed as long as it doesn't require a rooted phone (the resulting code must still be packaged as an app on the market).

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• SoundPool repeating issue for Samsung Galaxy S3

  - by Alaa Eldin

  I'm trying to play a background sound for my application, I use SoundPool class, my problem is that, sound plays well only when I set the loop parameter with zero value, but it doesn't work for any other value. My code for initialization is: soundpool = new SoundPool(4, AudioManager.STREAM_MUSIC, 0); soundsMap = new HashMap<Integer, Integer>(); soundsMap.put(1, soundpool.load(this, R.raw.soundfile_1, 1)); soundsMap.put(2, soundpool.load(this, R.raw.soundfile_2, 1)); my code for playing is soundpool.play(1, 0.9f, 0.9f, 1, -1, 1f); as I mentioned sound works when I put (0) instead of (-1) for the loop value, anyone has any idea why (-1) or any value other than (0) doesn't work (there is no output sound) ?

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• Open Source sound engine

  - by Steph Thirion

  When I started using SoundEngine (from CrashLanding and TouchFighter), I had read about a few people recommending not to use it, for it was, according to them, not stable enough. Still it was the only solution I knew of to play sounds with pitch and position control without learning C++ and OpenAL, so I ignored the warnings and went on with it. But now I'm starting to worry. The 2.2 SDK introduced AVFoundation. Using both SoundEngine from CrashLanding (for sounds) and AVAudioPlayer (for music), I found out SoundEngine behaves strangely when the only existing AVAudioPlayer is released (all sounds stop until a new AVAudioPlayer is initiated). Around the same time as the 2.2 SDK came out, the CrashLanding sample code was mysteriously removed from the ADC site. I'm worried there are more bad surprises to come. My question is, is anyone aware of an Open Source alternative to SoundEngine? Maybe even a C++ library that uses OpenAL?

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• How to record sound from a microphone in VB6?

  - by Clay Nichols

  We've been recording sound for over a decade using what seems like a very clunky method using the Winmm.dll and the MCIsendString. I've read that this doesn't set the recording quality value correctly (not sure if that article was ever true or is still true). I was wondering if there is any better way to record sound.

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• A way to enable a LaunchDaemon to output sound?

  - by Varun Mehta

  I have a small Foundation application that checks a website and plays a sound if it sees a certain value. This application successfully plays a sound when I run it as my user from the Terminal. I've configured this app to run as a LaunchDaemon, with the following plist: <?xml version="1.0" encoding="UTF-8"?> <!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd"> <plist version="1.0"> <dict> <key>Label</key> <string>org.myorg.appidentifier</string> <key>ProgramArguments</key> <array> <string>/Users/varunm/path/to/cli/application</string> </array> <key>KeepAlive</key> <true/> <key>RunAtLoad</key> <true/> </dict> </plist> When I have this service launched I can see it successfully read in and log values from the website, but it never generates any sound. The sound files are located in the same directory as the binary, and I use the following code: NSSound *soundToPlay = [[NSSound alloc] initWithContentsOfFile:@"sound.wav" byReference:NO]; [soundToPlay setDelegate:stopper]; [soundToPlay play]; while (g_keepRunning) { [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:1.0]]; } [soundToPlay setCurrentTime:0.0]; Is there any way to get my LaunchDaemon application to play sound? This machine gets run by different people, and sometimes has no one logged in, which is why I have to configure it as a LaunchDaemon.

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• Is it possible to detect when the system is recording a sound and then perform some action on Python

  - by Jorge

  I began learning Python a few days ago, and i was wondering about a practical use for a program. Then i came up with the following: if my brother is in his room recording himself playing guitar, a led plugged to the usb and wired so it's outside his door lights up, and then i'll know he's recording and i'll take care not to make any noises. The main questions are: How Python can detect any recording going on in the system? How would i interface with the usb so i can actually turn the led on?

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• How to visualize sound?

  - by user311461

  When music is playing how can i draw a wave?

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• can the python wave module accept StringIO object

  - by user368005

  i'm trying to use the wave module to read wav files in python. whats not typical of my applications is that I'm NOT using a file or a filename to read the wav file, but instead i have the wav file in a buffer. And here's what i'm doing import StringIO buffer = StringIO.StringIO() buffer.output(wav_buffer) file = wave.open(buffer, 'r') but i'm getting a EOFError when i run it... File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 493, in open return Wave_read(f) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 163, in __init__ self.initfp(f) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/wave.py", line 128, in initfp self._file = Chunk(file, bigendian = 0) File "/System/Library/Frameworks/Python.framework/Versions/2.5/lib/python2.5/chunk.py", line 63, in __init__ raise EOFError i know the StringIO stuff works for creation of wav file and i tried the following and it works import StringIO buffer = StringIO.StringIO() audio_out = wave.open(buffer, 'w') audio_out.setframerate(m.getRate()) audio_out.setsampwidth(2) audio_out.setcomptype('NONE', 'not compressed') audio_out.setnchannels(1) audio_out.writeframes(raw_audio) audio_out.close() buffer.flush() # these lines do not work... # buffer.output(wav_buffer) # file = wave.open(buffer, 'r') # this file plays out fine in VLC file = open(FILE_NAME + ".wav", 'w') file.write(buffer.getvalue()) file.close() buffer.close()

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• Correct way to Convert 16bit PCM Wave data to float

  - by fredley

  I have a wave file in 16bit PCM form. I've got the raw data in a byte[] and a method for extracting samples, and I need them in float format, i.e. a float[] to do a Fourier Transform. Here's my code, does this look right? I'm working on Android so javax.sound.sampled etc. is not available. private static short getSample(byte[] buffer, int position) { return (short) (((buffer[position + 1] & 0xff) << 8) | (buffer[position] & 0xff)); } ... float[] samples = new float[samplesLength]; for (int i = 0;i<input.length/2;i+=2){ samples[i/2] = (float)getSample(input,i) / (float)Short.MAX_VALUE; }

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• Voice Stress Analysis in Python

  - by mudder

  I'm interested in playing with this, anyone know of any code for this already existing? or what libraries would be useful to get started on something like this?

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• What does LAME text does in MP3 file?

  - by Dims

  I see here http://en.wikipedia.org/wiki/MP3 that MP3 file consists of MP3 headers interchanged with MP3 data. MP3 header consist of few bytes. But here is my MP3 file dump with ID3 tag cut. Header is highlighted with blue. You can see that "LAME3.96" text is highlighted with green. What does it does there? Is this a part of MP3 elementary stream? Or this is the part of some headers I didn't tag?

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• What exactly does raw microphone data represent?

  - by esperantist

  I'm using PyAudio, a PortAudio wrapper for Python. I'm getting data from a microphone. Data which is represented by a continuous stream of bytes divided into chunks (of a size determined by me). I've tried to plot the signal, assuming the bytes represent the current signal amplitude, but I get an interesting image that I can't easily describe. ^^ It seems to be composed of two waves, one shifted from the other. What exactly do the particular bytes represent, and how does this change when I'm recording only one channel, instead of two? Any explanations, suggestions, code snippets, anything, very welcome! (I'm new at this.) Thanks!

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