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• What language/API to use for a standalone live-input audio visualizer app?

  - by knuckfubuck

  I develop with Actionscript and was glad to see that AIR 2.0 was going to give access to mic input data. I planned to use this to create a visualizer set to the tempo of the incoming live audio. After doing a few days of google research it seems unlikely that it will be possible to analyze the data of the mic input in Flash/AIR. If anyone has ideas on how I can achieve this in AIR please let me know. (I'm open to workarounds.) That being said, I don't want to give up on the idea so I'm interested in suggestions for other language/API to use. My requirements for the app are: Run on OSX Two windows - one that can go fullscreen while the other(controller GUI) stays put Able to access live mic input data I've done reading on FFT and understand what needs to be done on the sound side so no need to help with that.

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• OpenAL - determine maximum sources

  - by Bill Kotsias

  Is there an API that allows you to define the maximum number of OpenAL "sources" allowed by the underlying sound hardware? Searching the internet, I found 2 recommendations : keep generating OpenAL sources till you get an error. However, there is a note in FreeSL (OpenAL wrapper) stating that this is "very bad and may even crash the library" assume you only have 16; why would anyone ever require more? (!) The second recommendation is even adopted by FreeSL. So, is there a common API to define the number of simultaneous "voices" supported? Thank you for your time, Bill

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• Speech recognition with Flash or Silverlight

  - by Sebastián Grignoli

  I'm developing a web user interface to enter some information that is not very complex but needs to be loaded in real time. I think that the application could make use of speech recognition to facilitate the task. Te core of the interface is being built with Javascript and jQuery, but can easily include a flash or silverlight component. I believe that´s probably the way to go... I don't need to recognize everything that the user says, but only a few prerecorded commands. Also, I don't want the user to click on a button to specify the begining and the end of the spoken command. It should be detected live. Is there anything that does this? I would be grateful if anyone tells me about a complete solution, free or commercial, as well as any advice on capturing a sound stream from the mic and process it with flash or sliverlight. Sebastian.-

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• SoundPool.load() and FileDescriptor from file

  - by Hans

  I tried using the load function of the SoundPool that takes a FileDescriptor, because I wanted to be able to set the offset and length. The File is not stored in the Ressources but a file on the storage card. Even though neither the load nor the play function of the SoundPool throw any Exception or print anything to the console, the sound is not played. Using the same code, but use the file path string in the SoundPool constructor works perfectly. This is how I have tried the loading (start equals 0 and length is the length of the file in miliseconds): FileInputStream fileIS = new FileInputStream(new File(mFile)); mStreamID = mSoundPool.load(fileIS.getFD(), start, length, 0); mPlayingStreamID = mSoundPool.play(mStreamID, 1f, 1f, 1, 0, 1f); If I would use this, it works: mStreamID = mSoundPool.load(mFile, 0); mPlayingStreamID = mSoundPool.play(mStreamID, 1f, 1f, 1, 0, 1f); Any ideas anyone? Thanks

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• Check if song is buffering in AS3

  - by SXMC

  I have the following piece of code: var song:Sound; var sndChannel:SoundChannel; var context:SoundLoaderContext = new SoundLoaderContext(2000); function songLoad():void { song.load(new URLRequest(songs[selected]),context); sndChannel = song.play(); } Now I want to be able to check if the song is buffering or not. Is there a way to do this? Or should I approach it differently? Thanks in advance!

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• iPhone: How many instances of AVAudioPlayer should I have for multiple sounds?

  - by foreyez

  So I'm using AvAudioPlayer to play multiple wav files. About 20 different sounds (each about 1 sec long), and you can think of each being played on a button press. Also I don't need them all to play simultaneously, i.e., one plays and you press another button to play another one (which stops the currently played one). What I'm wondering, should I have multiple instances of AVAudioPlayer (20 of them) and then preload the audio files, or should I just use one instance of AvAudioPlayer and each time a button is pressed, initialize the AvAudioPlayer with the sound url (or would this be too slow?) Thanks in advance!

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• Linux, C++ audio capturing (just microphone) library

  - by TheOm3ga

  I'm developing a musical game, it's like a singstar but instead of singing, you have to play the recorder. It's called oFlute, and it's still in early development stage. In the game, I capture the microphone input, then run a simple FFT analysis and compare the results to typical recorder's frequencies, thus getting the played note. At the beginning, the audio library I was using was RtAudio, but I don't remember why I switched to PortAudio, which is what I'm currently using. The problem is that, from time to time, either it crashes randomly or stops capturing, like if there were no sound coming from the microphone. My question is, what's the best option to capture microphone input on Linux? I just need to open, read, and close a flow of bytes from the microphone. I've been reading this guide, and (un)surprisingly it says: I don't think that PortAudio is very good API for Unix-like operating systems. So, what do you recommend me?

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• Comparing two speech sounds

  - by JessicaB

  I need to be able to determine if two sounds are very similar. The goal is to have a very limited vocabulary (10 or 15) of short one or two syllable words, then compare a captured sound to determine if it is one of those items with all the usual variability in environmental and capture conditions. The idea is that the user can issue a few simple commands by voice instead of keyboard or mouse. Does anyone know the best approach to this? I don't want to do full blown speech recognition, just something much more limited.

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• Using Pidgin 2.5.2 in Linux no sound is made for incoming messages (as it should)

  - by Kent

  Hi, I have a problem with Pidgin 2.5.2 in Linux (Ubuntu 8.10). When someone sends me a message no sound is played (the tray indicates a new message and blinks, that's it). Sounds play fine when I send someone a message. If I preview the Message received and Message sent sound events both of them do make a sound. Automatic and ALSA is what's working from the alternatives in Sound method selection. I include a screenshot containing a lot of relevant information: Screenshot (It's to big to fit nicely inline.)

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• Slideshow from excel file listing the caption, sound file and image file?

  - by Slabo

  Hello, I have excel files with the following header: Caption Sound: Location of sound file Image: Location of image file How can I make a slideshow from this? Each slide should show image, caption, and play sound automatically according to the excel list. I don't care what software I use, if I can get the job done. Total slides ~10,000. In case interested,this is review material for English second language students. Any help appreciated, Thanks

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• How to play the same Sound multiple times with overlap, using OpenAL or Finch?

  - by mystify

  Finch uses OpenAL. However, when I have an instance of Sound, and say -play, the sound plays. When I call -play multiple times one after another in a fast paced way, every -play makes the current sound playback of that sound stop and restart it. That's not what I want. Would I have to create multiple sources or buffers to get that working? Or would I just instantiate multiple Sounds with the same file?

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• How do i make a sound be able to hod long (read more please i need help)

  - by EstherBunny

  Ok what i need is help..haha ok im making a music app with a paino but i need to be able to hold the sound long or short. So like if i hold a button it will start playing the sound but if i let go it stops WITH A FADE so how ever long i hold the button down that is how long the sound will play (that is ntill the sound ends)

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• Assign programs permanently to different sound-outputs in Pulseaudio?

  - by Mood

  I want to assign Skype input and output to my USB-headset while the rest of my laptop uses the internal sound-card. This is an easy task with PulseAudio Volume control (pavucontrol). The only problem I have is every time a call is made I manually have to set the output and input for Skype to my USB-device . When I hang up, Skype disappears from Volume Control. It reappears again with the next call only this time the default sound-card is selected again. It shouldn’t be hard to let PulseAudio look or the USB-headset is connected when Skype audio comes is before selecting the default. The way to do it is obvious not through Volume Control.

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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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• My sound stopped working today, how can I fix it?

  - by Oli

  This seems to be a problem with pulseaudio. I was logged in over VNC on my phone and started playing a video this caused X to crash (as sometimes happens). I restarted and suddenly the sound doesn't work. I have a Intel HDA/Realtek ALC889 00:1b.0 Audio device: Intel Corporation 82801JI (ICH10 Family) HD Audio Controller alsamixer is detecting this just fine. PulseAudio doesn't detect this alsa device so is using auto_null as the default sink (logs below). When I properly kill PulseAudio (tell it not to auto-start) direct ALSA communication with the sound card works just fine. speaker-test, for example, works. So the hardware and ALSA layers are fine IMO. In the logs, it seems that the card might be "busy" but I really don't know how or why it would be now (and never before). Is there an ALSA lock file somewhere that it still there because of my crash? I just ran sudo fuser /dev/snd/* and saw this: oli@bert:~$ sudo fuser /dev/snd/* /dev/snd/controlC0: 1884 /dev/snd/pcmC0D0c: 1884m /dev/snd/timer: 1884 A look at the process list (ps aux | grep 1884) tells me process 1884 is arecord -c 1 -f S16_LE -r 8000 -t raw. No idea what this is or why it's running. When I try and kill arecord (as root), it just respawns and rebinds on the hardware. I'm in a very annoying situation where I don't know what is going on and don't know how to find out. I'm open to all suggestions to get this working again. Fire away. And here's what I get when I stop PA auto-loading, kill it and then start it with -vvvv. oli@bert:~$ pulseaudio -vvvvv I: main.c: setrlimit(RLIMIT_NICE, (31, 31)) failed: Operation not permitted I: main.c: setrlimit(RLIMIT_RTPRIO, (9, 9)) failed: Operation not permitted D: core-rtclock.c: Timer slack is set to 50 us. D: core-util.c: RealtimeKit worked. I: core-util.c: Successfully gained nice level -11. I: main.c: This is PulseAudio 0.9.21-63-gd3efa-dirty D: main.c: Compilation host: x86_64-pc-linux-gnu D: main.c: Compilation CFLAGS: -g -O2 -g -Wall -O3 -Wall -W -Wextra -pipe -Wno-long-long -Winline -Wvla -Wno-overlength-strings -Wunsafe-loop-optimizations -Wundef -Wformat=2 -Wlogical-op -Wsign-compare -Wformat-security -Wmissing-include-dirs -Wformat-nonliteral -Wold-style-definition -Wpointer-arith -Winit-self -Wdeclaration-after-statement -Wfloat-equal -Wmissing-prototypes -Wstrict-prototypes -Wredundant-decls -Wmissing-declarations -Wmissing-noreturn -Wshadow -Wendif-labels -Wcast-align -Wstrict-aliasing=2 -Wwrite-strings -Wno-unused-parameter -ffast-math -Wp,-D_FORTIFY_SOURCE=2 -fno-common -fdiagnostics-show-option D: main.c: Running on host: Linux x86_64 2.6.38-rc3 #1 SMP Tue Feb 1 10:53:04 GMT 2011 D: main.c: Found 8 CPUs. I: main.c: Page size is 4096 bytes D: main.c: Compiled with Valgrind support: no D: main.c: Running in valgrind mode: no D: main.c: Running in VM: no D: main.c: Optimised build: yes D: main.c: All asserts enabled. I: main.c: Machine ID is 8310740c4729ef474fe5ecec4bbf5a6b. I: main.c: Session ID is 8310740c4729ef474fe5ecec4bbf5a6b-1297338553.571075-1050119523. I: main.c: Using runtime directory /home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-runtime. I: main.c: Using state directory /home/oli/.pulse. I: main.c: Using modules directory /usr/lib/pulse-0.9.21/modules. I: main.c: Running in system mode: no I: main.c: Fresh high-resolution timers available! Enjoy ol' chap! I: cpu-x86.c: CPU flags: CMOV MMX SSE SSE2 SSE3 SSSE3 SSE4_1 SSE4_2 I: svolume_mmx.c: Initialising MMX optimized functions. I: remap_mmx.c: Initialising MMX optimized remappers. I: svolume_sse.c: Initialising SSE2 optimized functions. I: remap_sse.c: Initialising SSE2 optimized remappers. I: sconv_sse.c: Initialising SSE2 optimized conversions. D: memblock.c: Using shared memory pool with 1024 slots of size 64.0 KiB each, total size is 64.0 MiB, maximum usable slot size is 65472 D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes.tdb' I: module-device-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-device-volumes'. I: module.c: Loaded "module-device-restore" (index: #0; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes.tdb' I: module-stream-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-stream-volumes'. I: module.c: Loaded "module-stream-restore" (index: #1; argument: ""). D: database-tdb.c: Opened TDB database '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database.tdb' I: module-card-restore.c: Sucessfully opened database file '/home/oli/.pulse/8310740c4729ef474fe5ecec4bbf5a6b-card-database'. I: module.c: Loaded "module-card-restore" (index: #2; argument: ""). I: module.c: Loaded "module-augment-properties" (index: #3; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards. I: module.c: Loaded "module-udev-detect" (index: #4; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-bluetooth-discover.so': success D: dbus-util.c: Successfully connected to D-Bus system bus ba7c9a1f90b3d49d930bca2100000015 as :1.62 D: bluetooth-util.c: dbus: interface=org.freedesktop.DBus, path=/org/freedesktop/DBus, member=NameAcquired D: bluetooth-util.c: Bluetooth daemon is apparently not available. I: module.c: Loaded "module-bluetooth-discover" (index: #5; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-esound-protocol-unix.so': success I: module.c: Loaded "module-esound-protocol-unix" (index: #6; argument: ""). I: module.c: Loaded "module-native-protocol-unix" (index: #7; argument: ""). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-gconf.so': success I: module.c: Loaded "module-gconf" (index: #8; argument: ""). I: module-default-device-restore.c: Saved default sink 'auto_null' not existant, not restoring default sink setting. I: module-default-device-restore.c: Saved default source 'auto_null.monitor' not existant, not restoring default source setting. I: module.c: Loaded "module-default-device-restore" (index: #9; argument: ""). I: module.c: Loaded "module-rescue-streams" (index: #10; argument: ""). D: module-always-sink.c: Autoloading null-sink as no other sinks detected. I: sink.c: Created sink 0 "auto_null" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: sink.c: device.description = "Dummy Output" I: sink.c: device.class = "abstract" I: sink.c: device.icon_name = "audio-card" D: core-subscribe.c: Dropped redundant event due to change event. I: source.c: Created source 0 "auto_null.monitor" with sample spec s16le 6ch 44100Hz and channel map front-left,front-left-of-center,front-center,front-right,front-right-of-center,rear-center I: source.c: device.description = "Monitor of Dummy Output" I: source.c: device.class = "monitor" I: source.c: device.icon_name = "audio-input-microphone" D: module-null-sink.c: Thread starting up I: module.c: Loaded "module-null-sink" (index: #11; argument: "sink_name=auto_null sink_properties='device.description="Dummy Output"'"). I: module.c: Loaded "module-always-sink" (index: #12; argument: ""). I: module.c: Loaded "module-intended-roles" (index: #13; argument: ""). D: module-suspend-on-idle.c: Sink auto_null becomes idle, timeout in 5 seconds. I: module.c: Loaded "module-suspend-on-idle" (index: #14; argument: ""). I: client.c: Created 0 "ConsoleKit Session /org/freedesktop/ConsoleKit/Session1" D: module-console-kit.c: Added new session /org/freedesktop/ConsoleKit/Session1 I: module.c: Loaded "module-console-kit" (index: #15; argument: ""). I: module.c: Loaded "module-position-event-sounds" (index: #16; argument: ""). D: dbus-util.c: Successfully connected to D-Bus session bus efbffc6788fad56cfd64d40c00000018 as :1.182 D: main.c: Got org.pulseaudio.Server! I: main.c: Daemon startup complete. I: client.c: Created 1 "Native client (UNIX socket client)" I: client.c: Created 2 "Native client (UNIX socket client)" D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: protocol-native.c: Protocol version: remote 16, local 16 I: protocol-native.c: Got credentials: uid=1000 gid=1000 success=1 D: protocol-native.c: SHM possible: yes D: protocol-native.c: Negotiated SHM: yes D: module-augment-properties.c: Looking for .desktop file for gnome-volume-control-applet D: module-augment-properties.c: Looking for .desktop file for gnome-settings-daemon D: core-subscribe.c: Dropped redundant event due to change event. I: module-suspend-on-idle.c: Sink auto_null idle for too long, suspending ... D: sink.c: Suspend cause of sink auto_null is 0x0004, suspending Note the one section that seems to find the hardware but says it's busy (no idea if this is relevant). D: cli-command.c: Checking for existance of '/usr/lib/pulse-0.9.21/modules/module-udev-detect.so': success D: module-udev-detect.c: /dev/snd/controlC0 is accessible: yes D: module-udev-detect.c: /devices/pci0000:00/0000:00:1b.0/sound/card0 is busy: yes I: module-udev-detect.c: Found 1 cards.

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• flex 4: swfloader - how to mute game completly

  - by ufk

  Hiya. Ive read some answers here regarding muting swfloader volume but none of the examples would work in flex 4. I tried doinf the following: this._swfGame.source=url; this._swfGame.soundTransform = new SoundTransform(0.0); this would shut down the volume of the preloader, but when the game starts the volume is back to normal. i tried adding the following to the previous code: this._swfGame.addEventListener(Event.COMPLETE,this._configSwf); private function _configSwf(event:Event):void { this._swfGame.removeEventListener(Event.COMPLETE, _configSwf); var soundTransform:SoundTransform = new SoundTransform(0.0); // TODO: set proper volume this._swfGame.soundTransform = soundTransform; } but i got the same results. any ideas? thanks!

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• c++ FFT Beat detection library?

  - by mokaschitta

  Hi, I am currently looking around for a good allround beat detection library / source code in C++ since I found it really hard to achieve satisfying results with the beat detection code I wrote myself using this tutorial: http://www.gamedev.net/reference/programming/features/beatdetection/ It's especially really hard if you want to make it work with any kind of music so I was wondering if there is something usable out there allready? Thanks!

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• C# DirectSound - Capture buffers not continuous

  - by Wizche

  Hi, I'm trying to capture raw data from my line-in using DirectSound. My problem is that, from a buffer to another the data are just inconsistent, if for example I capture a sine I see a jump from my last buffer and the new one. To detected this I use a graph widget to draw the first 500 elements of the last buffer and the 500 elements from the new one: Snapshot I initialized my buffer this way: format = new WaveFormat { SamplesPerSecond = 44100, BitsPerSample = (short)bitpersample, Channels = (short)channels, FormatTag = WaveFormatTag.Pcm }; format.BlockAlign = (short)(format.Channels * (format.BitsPerSample / 8)); format.AverageBytesPerSecond = format.SamplesPerSecond * format.BlockAlign; _dwNotifySize = Math.Max(4096, format.AverageBytesPerSecond / 8); _dwNotifySize -= _dwNotifySize % format.BlockAlign; _dwCaptureBufferSize = NUM_BUFFERS * _dwNotifySize; // my capture buffer _dwOutputBufferSize = NUM_BUFFERS * _dwNotifySize / channels; // my output buffer I set my notifications one at half the buffer and one at the end: _resetEvent = new AutoResetEvent(false); _notify = new Notify(_dwCapBuffer); bpn1 = new BufferPositionNotify(); bpn1.Offset = ((_dwCapBuffer.Caps.BufferBytes) / 2) - 1; bpn1.EventNotifyHandle = _resetEvent.SafeWaitHandle.DangerousGetHandle(); bpn2 = new BufferPositionNotify(); bpn2.Offset = (_dwCapBuffer.Caps.BufferBytes) - 1; bpn2.EventNotifyHandle = _resetEvent.SafeWaitHandle.DangerousGetHandle(); _notify.SetNotificationPositions(new BufferPositionNotify[] { bpn1, bpn2 }); observer.updateSamplerStatus("Events listener initialization complete!\r\n"); And here is how I process the events. /* Process thread */ private void eventReceived() { int offset = 0; _dwCaptureThread = new Thread((ThreadStart)delegate { _dwCapBuffer.Start(true); while (isReady) { _resetEvent.WaitOne(); // Notification received /* Read the captured buffer */ Array read = _dwCapBuffer.Read(offset, typeof(short), LockFlag.None, _dwOutputBufferSize - 1); observer.updateTextPacket("Buffer: " + count.ToString() + " # " + read.GetValue(read.Length - 1).ToString() + " # " + read.GetValue(0).ToString() + "\r\n"); /* Print last/new part of the buffer to the debug graph */ short[] graphData = new short[1001]; Array.Copy(read, graphData, 1000); db.SetBufferDebug(graphData, 500); observer.updateGraph(db.getBufferDebug()); offset = (offset + _dwOutputBufferSize) % _dwCaptureBufferSize; /* Out buffer not used */ /*_dwDevBuffer.Write(0, read, LockFlag.EntireBuffer); _dwDevBuffer.SetCurrentPosition(0); _dwDevBuffer.Play(0, BufferPlayFlags.Default);*/ } _dwCapBuffer.Stop(); }); _dwCaptureThread.Start(); } Any advise? I'm sure I'm failing somewhere in the event processing, but I cant find where. I had developed the same application using the WaveIn API and it worked well. Thanks a lot...

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• Play any audio for given time

  - by Dipen

  I want to play any file for 6 seconds. Also suppose the audio is bigger then 6 sec the application will play only for 6 sec.and if it is less then 6 sec then play continuously. So is there any inbuilt option from any framework?

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• How to make a simple voice recorder in pcm format?

  - by XBasic3000

  i created a simple voice recording using wav format. my problem is i wanted it to save as PCM format without using comercial components. i have MSACM.pas but i dont know how to use it. My compiler is delphi.

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• Flash computeSpectrum() unsynchronized with audio

  - by sold

  I am using Flash's (CS4, AS3) SoundMixer.computeSpectrum to visualize a DFT of what supposed to be, according to the docs, whatever is currently being played. However, there is a considerable delay between the audio and the visualization (audio comes later). It seems that computeSpectrum captures whatever is on it's way to the buffer, and not to the speakers. Any cure for this?

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• play two sounds simultaneously iphone sdk

  - by Asaf Greene

  I am trying to make a small music app on the iphone. I want to have an octave a piano which will respond to touches and play the key or keys that the user touches. How would i be able to get two or more sounds to play at the same time so it sounds like a chord? I tried using AVFoundation but the two sounds just play one after the other.

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• What is the easiest way to read wav-files using Python [summary]?

  - by Roman

  I want to use Python to access a wav-file and write its content in a form which allows me to analyze it (let's say arrays). I heard that "audiolab" is a suitable tool for that (it transforms numpy arrays into wav and vica versa). I have installed the "audiolab" but I had a problem with the version of numpy (I could not "from numpy.testing import Tester"). I had 1.1.1. version of numpy. I have installed a newer version on numpy (1.4.0). But then I got a new set of errors: Traceback (most recent call last): File "test.py", line 7, in import scikits.audiolab File "/usr/lib/python2.5/site-packages/scikits/audiolab/init.py", line 25, in from pysndfile import formatinfo, sndfile File "/usr/lib/python2.5/site-packages/scikits/audiolab/pysndfile/init.py", line 1, in from _sndfile import Sndfile, Format, available_file_formats, available_encodings File "numpy.pxd", line 30, in scikits.audiolab.pysndfile._sndfile (scikits/audiolab/pysndfile/_sndfile.c:9632) ValueError: numpy.dtype does not appear to be the correct type object I gave up to use audiolab and thought that I can use "wave" package to read in a wav-file. I asked a question about that but people recommended to use scipy instead. OK, I decided to focus on scipy (I have 0.6.0. version). But when I tried to do the following: from scipy.io import wavfile x = wavfile.read('/usr/share/sounds/purple/receive.wav') I get the following: Traceback (most recent call last): File "test3.py", line 4, in <module> from scipy.io import wavfile File "/usr/lib/python2.5/site-packages/scipy/io/__init__.py", line 23, in <module> from numpy.testing import NumpyTest ImportError: cannot import name NumpyTest So, I gave up to use scipy. Can I use just wave package? I do not need much. I just need to have content of wav-file in human readable format and than I will figure out what to do with that.

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• OpenAL device, buffer and context relationship

  - by Markus

  I'm trying to create an object oriented model to wrap OpenAL and have a little problem understanding the devices, buffers and contexts. From what I can see in the Programmer's Guide, there are multiple devices, each of which can have multiple contexts as well as multiple buffers. Each context has a listener, and the alListener*() functions all operate on the listener of the active context. (Meaning that I have to make another context active first if I wanted to change it's listener, if I got that right.) So far, so good. What irritates me though is that I need to pass a device to the alcCreateContext() function, but none to alGenBuffers(). How does this work then? When I open multiple devices, on which device are the buffers created? Are the buffers shared between all devices? What happens to the buffers if I close all open devices? (Or is there something I missed?)

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• monotouch play music when device is locked

  - by Ali Shafai

  I'm trying to make my monotouch app continue playing when the device is locked, I found this snippet in ObjC, was wondering if mt already has bindings for it or not. AudioSessionInitialize (NULL,NULL,interruptionListenerCallback,self); UInt32 sessionCategory = kAudioSessionCategory_MediaPlayback; AudioSessionSetProperty(kAudioSessionProperty_AudioCategory, sizeof(sessionCategory), &sessionCategory);

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