community!
I use ZEBRA MC3300/android 8.1
I would really appreciate if you'll help me with one question.
The main reason I want to do it is to reduce amount of human error possibly can happen.
When VELOCITY app is on and a worker scans a barcode, there is usually plays a beep sound.
But the problem is that if a worker scans wrong barcode or an object, that is not counted in store, usually the same sound plays.
So I tried to edit this via console.
so, basically, the sound I've added works fine, but an error beep isn't playing when I try to scan any random(not our) object's barcode.
Is there a solution for my problem?
Sorry if my code looks too silly, I'm just trying to improve my workspace :(
Thanks in advance!
function replaceBeep(beepType,soundFile) {
function onBeep(event) {
if(event.type == beepType)
{
Device.beepPlayFile(soundFile);
event.eventHandled = true;
}
}
WLEvent.on("Beep", onBeep);
Device.errorBeep(1000, 300, 50, 0.5);
}
WLEvent.registerScope("session",replaceBeep,null,["2","error.wav"]);`
expecting to play separate beep sounds for wrong and right scan.
Related
I am creating a game using Libgdx. I have a lot of small sounds files in the MP3-format and since it is so many I do not preload them. I only load the sound I want to play when it is to be used, like this:
actorSound = Gdx.audio.newSound(Gdx.files.internal(sound));
The code above works great, but the rest of my sounds do not unfortunately. The actor above has its own class and plays a different sound every time it is touched.
When I try to play sounds in my Gamescreen I get the following error:
AUDIO_OUTPUT_FLAG_FAST denied by client
All the audiofiles have the same properties and have been recorded using the same microphone & Audacity. The files are all 44100Hz and only a few kb in size each.
I wonder why the sounds the Actor plays work and the other sounds do not?
I decided to try to change the non-working sounds to music instead and now they play fine - for a little while that is. I can play a full game, return to the menu and start a new game again. The second time I start a game I only get 3 sounds from the Gamescreen and then it is silent except for the sounds from the actor. The error that appears looks like this:
E/MediaPlayer: Error (1,-19)
I load the Music just the same way as the Sound:
gameSound = Gdx.audio.newMusic(Gdx.files.internal(soundeffect));
I have looked into the two errors by reading posts like these:
AUDIO_OUTPUT_FLAG_FAST denied by client
Mediaplayer error (-19,0) after repeated plays
But I'm not sure what to do or change to solve my problem. I would prefer Sound if that is possible, but Music is an acceptable workaround...
When it comes to Music it is probably as suggested in the URL, that I do not release the media players. I am not sure how to do that?
When I leave the GameScreen for another screen, like the MenuScreen or RewardScreen, I dispose Music first. The other screens use Music as well and the sounds are loaded when needed. When I change back to the GameScreen I dispose again and then start a new game...
Any ideas or suggestions? Any help is greatly appreciated.
I added AssetManager as suggested, and I now have a loding screen that loads all the sounds. I load them as sounds and not music, which is how I prefer it.
The sounds work well in the actual game but once I get to the reward screen, only the first sound plays and after that the app crashes with the following error:
06-02 07:47:09.774 6208-6282E/AndroidRuntime: FATAL EXCEPTION: GLThread 2935
Process: PID: 6208
java.lang.NullPointerException: Attempt to read from field 'com.badlogic.gdx.assets.AssetManager
Assets.Assets.manager' on a null object reference
at DelayedSounds(RewardsScreen.java:538)
at RewardsScreen.Update(RewardsScreen.java:567)
at Screens.RewardsScreen.render(RewardsScreen.java:577)
at Game.render(Game.java:46)
at com.badlogic.gdx.backends.android.AndroidGraphics.onDrawFrame(AndroidGraphics.java:459)
at android.opengl.GLSurfaceView$GLThread.guardedRun(GLSurfaceView.java:1562)
at android.opengl.GLSurfaceView$GLThread.run(GLSurfaceView.java:1262)
06-02 07:47:13.823 6208-6208 E/AndroidGraphics: waiting for pause
synchronization took too long; assuming deadlock and killing
The DelayedSounds method looks like this:
public void DelayedSounds(){
timer = timer + Gdx.graphics.getDeltaTime();
if(playitem == true && timer > 2){
itemsound = "vinster/" + item + ".mp3";
assets.manager.get(itemsound, Sound.class).play(volume);
//sound = Gdx.audio.newMusic(Gdx.files.internal(itemsound));
//sound.setVolume(volume);
//sound.play();
playitem = false;
}
if(playkeep == true && time > 3){
assets.manager.get("dialog/VINSTEN.mp3", Sound.class).play(volume);
//sound = Gdx.audio.newMusic(Gdx.files.internal("prizes/prize.mp3"));
//sound.setVolume(volume);
//sound.play();
playkeep = false;
}
}
As can be seen I use AssetManager now, and have commented out my old code for testing purposes. I define AssetManager in my main class and then pass it around to all other classes that uses sounds.
The RewardsScreen will only play the first sound and then it crashes on a NULL object reference as it seems.
If I change my code back to using Music in the RewardsScreen it works fine (see the code that is commmented out)
The sound I try to play is exactly the same in both cases. Assets class that handles the loading of all my assets has the sounds included and since I still get a NULL object I assume one or more items fails to load?
I search the logs and find this where it loads the sounds:
06-02 08:05:09.360 9844-9888/E/WVMExtractor: Failed to open libwvm.so: dlopen failed: library "libwvm.so" not found
06-02 08:05:09.395 9844-9888/ E/NdkMediaExtractor: sf error code: -1010
06-02 08:05:09.395 9844-9888/ E/SoundPool: Unable to load sample
Maybe this is related to my problem?
I load all the sounds in the same manner and most definitely seem to work, the loading is the regular:
manager.load("prizes/cash.mp3", Sound.class);
I still find the AUDIO_OUTPUT_FLAG_FAST denied by client in my logs but now the sounds are playing instead of being rejected.
Any more ideas about how to solve this?
Your link having enough information for your bug.
It's not good way to create Resource instance each time in Game, Create once and user All over your game, if possible use AssetManager.
Like create Music instance in onCreate() method of your game.
gameSound = Gdx.audio.newMusic(Gdx.files.internal(soundeffect));
Music having play(), resume() and many other helpful methods.
You can also take a look of this, it may be helpful.
I have a watchOS3 workout app that uses haptic notifications. It setup correctly to run a workout session and haptics work when running in the background. However, if bluetooth headphones are connected to the apple watch, then you only get the haptic vibration OR the audio chime for the haptic, depending on whether the app is currently showing on the watch face or running in background.
Here's how I'm playing the haptic:
WKInterfaceDevice.current().play(.notification)
Here are the details:
Apple Watch Nike+ Unpaired to Headphones: Haptic and chime sound activate regardless of whether the watch face is on or off. Chime sound is loud and clear.
Apple Watch Nike+ Paired to Bluetooth Headphones:
Haptic only active if watch face is on, audio chime is off. Audio chime is on when watch face is off, haptic is off. Chime sound is loud and clear.
I tested the app pairing the Apple Watch separately with the Platronic Backbeat Go 2 (released 7/2013) and the Bose QuietControl 30 (released 10/2016). The results were the same.
Anyone know if this is a limitation of watchOS 3, a bug, or is there something else I need to be doing?
Thanks,
Jeff
There's another factor to consider, which is whether the main Watch mute switch is on or off.
I found the following to be a suitable workaround.
When you want your audio/haptic to be noticed while music is playing on Bluetooth Headphones, have this AVAudioSession category set:
AVAudioSession.sharedInstance().setCategory(AVAudioSessionCategoryPlayback, with: .duckOthers)
Then, when you're done with your audio/haptic, reset the AVAudioSession back to:
AVAudioSession.sharedInstance().setCategory(AVAudioSessionCategoryPlayback, with: .mixWithOthers)
I have this helper class to help manage the states:
import AVFoundation
enum AudioPlaybackState {
case playback
case playbackDuckOthers
}
class AVAudio {
private init() {} // strictly a helper class
static func setAudioState(_ state: AudioPlaybackState) {
DispatchQueue.main.async {
do {
deactivateAudioSession()
switch state {
case .playback:
try AVAudioSession.sharedInstance().setCategory(AVAudioSessionCategoryPlayback, with: .mixWithOthers)
case .playbackDuckOthers:
try AVAudioSession.sharedInstance().setCategory(AVAudioSessionCategoryPlayback, with: .duckOthers)
}
activateAudioSession()
} catch {}
}
}
static func deactivateAudioSession() {
activateAudioSession(false)
}
private static func activateAudioSession(_ value: Bool=true) {
do {
try AVAudioSession.sharedInstance().setActive(value)
} catch {}
}
}
Then, I can switch quickly by: AVAudio.setAudioState(.playbackDuckOthers)
The behavior documented in the question is the the way its supposed to be, according to Apple. The given reason is because of the delay between the haptic tap and the haptic audio when using bluetooth headphones. I'm hoping this behavior changes in the future...
Can anyone tell me the best approach to playing single-tone, audio (.mp3) files in a Windows Phone 8 app? Think of a piano app, where each key would represent a button, and each button would play a different tone.
I'm looking for the most efficient way to go about this - I've got 8 different buttons that need to play a different tone when tapped.
I tried using the MediaElement:
MediaElement me;
public MainPage()
{
InitializeComponent();
me = new MediaElement();
me.AutoPlay = false;
me.Source = new Uri("/Sounds/Sound1.mp3", UriKind.Relative);
btnPlay.Click += btnPlay_Click;
}
private void btnPlay_Click(object sender, EventArgs e)
{
me.Play();
}
But nothing happens, either in the emulator or on a device (testing w/ a Lumia 822). Am I doing something wrong here? It seems like it should be pretty simple. Or would using MediaElement even be the best thing to use for my scenario?
Would this fall under the Background Audio category? I've read through this example but it seems overkill for what I want to do.
I've also read about using XNA's SoundEffect to do the job, but then I'd have to convert my .mp3 files to .wav (which isn't necessarily a problem, but I'd rather not go through that if I don't need to).
Can anyone tell me either what I'm doing wrong in my example above or guide me to a better solution for playing quick <1s audio tones?
I had this problem before with MediaElement not playing audio files. After many attempts I found out that it only plays if it defined in the xaml and AutoPlay is set to true.
Try defining it in the xaml or you can just add it to your LayoutRoot.
var me = new MediaElement();
LayoutRoot.Children.Add(me);
me.AutoPlay = true;
me.Source = new Uri("Sound/1.mp3", UriKind.Relative);
I have had good luck just doing this piece of code in my app. But it may not work as well in your context, give it a whirl though.
mediaElement.Source = new Uri("/Audio/" + songID.ToString() + ".mp3", UriKind.Relative);
mediaElement.Play();
Ok, I hope I don't mess this up, I have had a look for some answers but can't find anything. I am trying to make a simple sampler in openframeworks using the FMOD sound player in 3D mode. I can make a single instance work fine (recording a new file using libsndfilerecorder and then playing it back and moving it in surround.
However I want to have 8 layers of looping audio that I can record and replace one layer at a time in a live show. I get a lot of problems as soon as I have more than 1 layer.
The first part of my question relates to the FMOD 3D modes, it is listener relative, so I have to define the position of my listener for every sound (I would prefer to have head relative mode but I cannot make this work at all. Again this works fine when I am using a single player but with multiple players only the last listener I update actually works.
The main problem I have is that when I use multiple players I get distortion, and often a mix of other currently playing sounds (even when the microphone cannot hear them) in my new recordings. Is there an incompatability with libsndfilerecorder and FMOD?
Here I initialise the players
for (int i=0; i<CHANNEL_COUNT; i++) {
lvelocity[i].set(1, 1, 1);
lup[i].set(0, 1, 0);
lforward[i].set(0, 0, 1);
lposition[i].set(0, 0, 0);
sposition[i].set(3, 3, 2);
svelocity[i].set(1, 1, 1);
//player[1].initializeFmod();
//player[i].loadSound( "1.wav" );
player[i].setVolume(0.75);
player[i].setMultiPlay(true);
player[i].play();
setupHold[i]==false;
recording[i]=false;
channelHasFile[i]=false;
settingOsc[i]=false;
}
When I am recording I unload the file and make sure the positions of the player that is not loaded are not updating.
void fmodApp::recordingStart( int recordingId ){
if (recording[recordingId]==false) {
setupHold[recordingId]=true; //this stops the position updating
cout<<"Start recording Channel " + ofToString(recordingId+1)+" setup hold is true \n";
pt=getDateName() +".wav";
player[recordingId].stop();
player[recordingId].unloadSound();
audioRecorder.setup(pt);
audioRecorder.setFormat(SF_FORMAT_WAV | SF_FORMAT_PCM_16);
recording[recordingId]=true; //this starts the libSndFIleRecorder
}
else {
cout<<"Channel" + ofToString(recordingId+1)+" is already recording \n";
}
}
And I stop the recording like this.
void fmodApp::recordingEnd( int recordingId ){
if (recording[recordingId]=true) {
recording[recordingId]=false;
cout<<"Stop recording" + ofToString(recordingId+1)+" \n";
audioRecorder.finalize();
audioRecorder.close();
player[recordingId].loadSound(pt);
setupHold[recordingId]=false;
channelHasFile[recordingId]=true;
cout<< "File recorded channel " + ofToString(recordingId+1) + " file is called " + pt + "\n";
}
else {
cout << "Sorry track" + ofToString(recordingId+1) + "is not recording";
}
}
I am careful not to interrupt the updating process but I cannot see where I am going wrong.
Many Thanks
to deal with the distortion, i think you will need to lower the volume of each channel on playback, try setting the volume to 1/8 of the max volume. there isn't any clipping going on so if the sum of sounds > 1.0f you will clip and it will sound bad.
to deal with crosstalk when recording: i guess you have some sort of feedback going on with the output, ie the output sound is being fed back into the input channel, probably by the operating system. if you run another app that makes sound do you also get that in your recording as well? if so then that is probably your problem.
if it works with one channel, try it with just 2, instead of jumping straight up to 8 channels.
in general i would try to abstract out the playback/record logic and soundPlayer/recorder into a separate class. you have a couple of booleans there and it's really easy to make mistakes with >1 boolean. is there any way you can replace the booleans with an enum or an integer state variable?
EDIT: I didn't see the date on your question :D Suppose you managed to do it by now. Maybe it helps somebody else..
I'm not sure if I can answer everything of your question, but I can share how I've worked with 3D sound in FMOD. I haven't worked with recording though.
For my own application a user can place sounds in 3D space around himself. For this I only have one Listener and multiple Sounds. In your code you're making a listener for every sound, are you sure that is necessary? I would imagine that this causes the multiple listeners to pick up multiple sounds and output that to your soundcard. So from the second sound+listener, both listeners pick up both sounds? I'm not a 100% sure but it sounds plausible to me.
I made a class to create sound objects (and one listener). Then I use a vector to store the objects and move trough them to render them.
My class SoundBox basically holds all the necessary things for FMOD
Making a "SoundBox" object and adding it to my soundboxes vector:
SoundBox * box = new SoundBox(box_loc, box_rotation, box_color);
box->loadVideo(ofToDataPath(video_files[soundboxes.size()]));
box->loadSound(ofToDataPath(sound_files[soundboxes.size()]));
box->setVolume(1);
box->setMultiPlay(true);
box->updateSound(box_loc, box_vel);"
box->play();
soundboxes.push_back(box);
Constructor for the SoundBox. I use a similar constructor in the same class for the listener, but since the listener will always be at the origin for me, it doesn't take any arguments and just sets all the listener locations to 0. The constructor for the listener only gets called once, while the one for the Sound gets called whenever I want to make a new one. (don't mind the box_color. I'm drawing physical boxes in this case..):
SoundBox::SoundBox(ofVec3f box_location, ofVec3f box_rotation, ofColor box_color) {
_box_location = box_location;
_box_rotation = box_rotation;
_box_color = box_color;
sound_position.x = _box_location.x;
sound_position.y = _box_location.y;
sound_position.z = _box_location.z;
sound_velocity.x = 0;
sound_velocity.y = 0;
sound_velocity.z = 0;
Then I just use a for loop to loop trough them and play them if they're not playing. I also have some similar code to select them and move then around.
for(auto box = soundboxes.begin(); box != soundboxes.end(); box++){
if(!(*box)->getIsPlaying())
(*box)->play();
}
I really hoped this helped. I'm not a very experienced programmer but this is how I got FMOD with multiple sounds to work in OpenFrameworks and hope you can use some of it. I just dumped as much of my code as I could :D
My main suggestion is to make one listener instead of more. Also having a class for making the sounds is useful if you, for instance, want to relocate the sounds after the initial placement.
Hope it helps and good luck :)
is it possible to detect vibration on the iPhone? I'm trying to figure out how to detect when the user smacks a desk or table when the phone is sitting on it. I remember reading somewhere you could detect a smack on a wooden table using the mic and AVFoundation. Any ideas?
Thanks
if you go down the microphone route, something like this might do the trick:
//somewhere during setup call this line
[anAVAudioRecorder setMeteringEnabled:YES];
//then in a method used to poll the audioMeter
[anAVAudioRecorder updateMeters];
averagePower = [anAVAudioRecorder averagePowerForChannel:0];
if (averagePower > threshold ) {
[musicPlayerClass play];
}