I use this:
err = channel->setVolume(someVolumeBetween0and1);
Even if err is FMOD_OK, the volume doesn't change. Am I doing something wrong? Is there any way to change the volume for a sound(channel)? Is there other range for volume instead of [0, 1]?
Thanks!
EDIT: I use setVolume just after this:
err = soundSystem->playSound(FMOD_CHANNEL_FREE, sound, false, &channel);
I found something about volume in the FMOD manual:
"When a sound is played, it will use the sound's default frequency, volume, pan, levels and priority...
To change channel attributes before the sound is audible, start the channel paused by setting the paused flag to true, and calling the relevant channel based functions. Following that, unpause the channel with Channel::setPaused."
So, right code should like this:
err = soundSystem->playSound(FMOD_CHANNEL_FREE, sound, true, &channel);
err = channel->setVolume(someVolumeBetween0and1);
err = channel->setPaused(false);
or, you can try this one also:
err = soundSystem->playSound(FMOD_CHANNEL_FREE, sound, false, &channel);
err = channel->setVolume(someVolumeBetween0and1);
change the order of setVolume and playSound, it works in my project
I'm not 100% sure, but the sound may need to be stopped/paused before setting the volume to unlock it and resume playing the sound after the volume has been set.
Other things to check could be that the sound is being played on the correct channel (or that you are setting the volume on the correct channel). Is the sound locked? Is sound a part of a channelgroup that could be overwriting the volume (although FMOD docs say that channelgroups should scale, not overwrite)?
I had this issue using FMOD and went down a lot of these same roads. I tried pausing the sound. I wondered if it was because my sound was looped.
In the end it was entirely an application issue. I was not using the right channel object.
There is no need to pause in order to change volume. The point of playing the sound initially paused, then setting the volume to a non-default value, then unpausing the sound, is to avoid playing that sound briefly at default volume.
try this
result = channle->setPaused(false);//after you set volume
Related
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.
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.
As explained here, OffsetSampleProvider can be used in order to play a specific portion of an audio file. Like this:
AudioFileReader AudioReader = new AudioFileReader("x.wav");
OffsetSampleProvider OffsetProvider = New OffsetSampleProvider(AudioReader);
OffsetProvider.SkipOver = TimeSpan.FromSeconds(5);
OffsetProvider.Take = TimeSpan.FromSeconds(8);
myWaveOut.Init(OffsetProvider);
myWaveOut.Play();
The above example will play an audio for 8 seconds, starting at second 5. However, if I want to play it again, it will not play, unless I set the Position property of the AudioFileReader to 0, and re-create a new instance of OffsetSampleProvider from it. So I would like to know if I'm missing something, or this is the way that OffsetSampleProvider should be used (and if it does, do I have to free any resources related to it).
You could copy the code for OffsetSampleProvider and add a Reset method to it. I'd also avoid using SkipOver for performance reasons and just set the CurrentTime of the AudioFileReader to 5 seconds directly before you 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 :)
I made a streaming music player and it works fine in the foreground.
But in the background iOS4, it doesn't play the next song automatically. ( remote control works )
The reason is AudioQueueStart return -12985.
I already check the audio session. it just fine. I use AudioQueueStart when it start to play the music.
How can you remove AudioQueueStart?
- (void)play
{
[self setupAudioQueueBuffers]; // calcluate the size to use for each audio queue buffer, and calculate the // number of packets to read into each buffer
OSStatus status = AudioQueueStart(self.queueObject, NULL);
}
I read the answer in the web about the AudioQueueStart fail subject.
One thing to check is that the AudioSession is active first.
In my case, I had previously set the session to inactive between song changes before starting a new song:AudioSessionSetActive(false);
Once I removed this AudioQueueStart works just fine from the background.
In my experience, the -12985 message occurs because another app already has an audio session active when you try to start playback in your app. Options are to 1) instruct the user to close the other app, or 2) set mix mode (see kAudioSessionProperty_OverrideCategoryMixWithOthers).
The disadvantage of mix mode is if you depend on lock screen art or remote controls, they won't work with mix mode set.
I also faced with such problem week ago. I've spent two days to find solution and I found it. May be this link will help (it is official answer): http://developer.apple.com/library/ios/#qa/qa1668/_index.html
Make sure that you activate session from applicationDidEnterBackground task handler. Now my application can play sound in background.
See this.
You probably need to include the following:
[[UIApplication sharedApplication] beginReceivingRemoteControlEvents];
Towards the bottom there is a reiteration of the how important that line is. As it is not mentioned in any of the three main audio audio guides (AVFoundation, AudioSession, or AudioQueue) it can easily be missed.
I have the same problem.
I registry the AudioSessionInterruptionListener, pause the audio when phone call, resume it after the call end. but get -12985 error code when call AudioQueueStart to resume.
My solution is that I try to call AudioQueueStart after 0.02s.
I don't know the reason.
On iOS7, AudioQueueStart was returning '!int' ('tni!'), though i'm sure no one would be surprised to find that it's not documented in the docs or headers. It was the same issue, though, and the same fix (setting the audio session to active in the background task handler) worked for me.