I setted up a sound with multichannelsupport so now I need to delay the sound in each single speaker. How can I do this with FMODex? Is it possible to do that?
Thanks for helping me! :)
So I've got an answer to my question by myself and fmod.org. I have to use the FMOD_DSP_TYPE_DELAY. With this type I can set the delay for each channel up to 10 seconds. More informations could be found in the documentation from fmod.
~Update~
Some code for interested fmod users:
FMOD_System_CreateDSPByType(system, FMOD_DSP_TYPE_DELAY, &dspDelay);
FMOD_Channel_AddDSP(channel, dspDelay, 0);
FMOD_DSP_SetActive(dspDelay, true);
while(true) {
FMOD_DSP_SetParameter(dspDelay, FMOD_DSP_DELAY_CH0, delayLeft);
FMOD_DSP_SetParameter(dspDelay, FMOD_DSP_DELAY_CH1, delayRight);
Sleep(10);
FMOD_System_Update(system);
}
Related
I've been looking around Swift documentation to save an audio output from AVAudioEngine but I couldn't find any useful tip.
Any suggestion?
Solution
I found a way around thanks to matt's answer.
Here a sample code of how to save an audio after passing it through an AVAudioEngine (i think that technically it's before)
newAudio = AVAudioFile(forWriting: newAudio.url, settings: nil, error: NSErrorPointer())
//Your new file on which you want to save some changed audio, and prepared to be bufferd in some new data...
var audioPlayerNode = AVAudioPlayerNode() //or your Time pitch unit if pitch changed
//Now install a Tap on the output bus to "record" the transformed file on a our newAudio file.
audioPlayerNode.installTapOnBus(0, bufferSize: (AVAudioFrameCount(audioPlayer.duration)), format: opffb){
(buffer: AVAudioPCMBuffer!, time: AVAudioTime!) in
if (self.newAudio.length) < (self.audioFile.length){//Let us know when to stop saving the file, otherwise saving infinitely
self.newAudio.writeFromBuffer(buffer, error: NSErrorPointer())//let's write the buffer result into our file
}else{
audioPlayerNode.removeTapOnBus(0)//if we dont remove it, will keep on tapping infinitely
println("Did you like it? Please, vote up for my question")
}
}
Hope this helps !
One issue to solve:
Sometimes, your outputNode is shorter than the input: if you accelerate the time rate by 2, your audio will be 2 times shorter. This is the issue im facing for now since my condition for saving the file is (line 10)
if(newAudio.length) < (self.audioFile.length)//audiofile being the original(long) audio and newAudio being the new changed (shorter) audio.
Any help here?
Yes, it's quite easy. You simply put a tap on a node and save the buffer into a file.
Unfortunately this means you have to play through the node. I was hoping that AVAudioEngine would let me process one sound file into another directly, but apparently that's impossible - you have to play and process in real time.
Offline rendering Worked for me using GenericOutput AudioUnit. Please check this link, I have done mixing two,three audios offline and combine it to a single file. Not the same scenario but it may help you for getting some idea. core audio offline rendering GenericOutput
I need to play 2 sounds with 2 AVAudioPlayer objects at the same exact time... so I found this example on Apple AVAudioPlayer Class Reference (https://developer.apple.com/library/mac/#documentation/AVFoundation/Reference/AVAudioPlayerClassReference/Reference/Reference.html):
- (void) startSynchronizedPlayback {
NSTimeInterval shortStartDelay = 0.01; // seconds
NSTimeInterval now = player.deviceCurrentTime;
[player playAtTime: now + shortStartDelay];
[secondPlayer playAtTime: now + shortStartDelay];
// Here, update state and user interface for each player, as appropriate
}
What I don't understand is: why also the secondPlayer has the shorStartDelay?
Shouldn't it be without? I thought the first Player needed a 0.1 sec delay as it is called before the second Player... but in this code the 2 players have the delay...
Anyone can explain me if that is right and why?
Thanks a lot
Massy
If you only use the play method ([firstPlayer play];), firstPlayer will start before the second one as it will receive the call before.
If you set no delay ([firstPlayer playAtTime:now];), the firstPlayer will also start before de second one because firstPlayer will check the time at which it is supposed to start, and will see that it's already passed. Thus, it will have the same behaviour as when you use only the play method.
The delay is here to ensure that the two players start at the same time. It is supposed to be long enough to ensure that the two players receive the call before the 'now+delay' time has passed.
I don't know if I'm clear (English is not my native langage). I can try to be more clear if you have questions
Yeah what he said ^ The play at time will schedule both players to start at that time (sometime in the future).
To make it obvious, you can set "shortStartDelay" to 2 seconds and you will see there will be a two second pause before both items start playing.
Another tip to keep in mind here are that when you play/pause AVAudioPlayer they dont actually STOP at exactly the same time. So when you want to resume, you should also sync the audio tracks.
Swift example:
let currentDeviceTime = firstPlayer.deviceCurrentTime
let trackTime = firstPlayer.currentTime
players.forEach {
$0.currentTime = trackTime
$0.play(atTime: currentDeviceTime + 0.1)
}
Where players is a list of AVAudioPlayers and firstPlayer is the first item in the array.
Notice how I am also resetting the "currentTime" which is how many seconds into the audio track you want to keep playing. Otherwise every time the user plays/pauses the track they drift out of sync!
I am initializing FMOD with 32 channels and playing short samples (1 second) with the following code:
result = system->init(32, FMOD_INIT_NORMAL , NULL);
// here I load the sounds //
result = system->playSound(FMOD_CHANNEL_FREE, grid[_sound], false, &channel);
It works as intended, overlapping sounds, but now I realized that when I have played 32 samples (not at the same time), only one sound can be played at a time. It looks like FMOD_CHANNEL_FREE behaves like an incremental counter and when it hits 32, it stays there, stopping the last sound while it's still playing to play the new one.
Do I have to remove sounds when they have stopped playing? How? I feel like I am missing something basic
Thanks!
Marc
I had the same problem. Turns out that I forgot to call system->update() every frame. Once I put that in, it worked fine.
It sounds like the channels are still playing (but silent), can you check Channel::isPlaying and see if they are still going?
Perhaps post some more of your code if that doesn't help.
can u verify that u initializing fmod system with more than one max channels?
try to use following code for init your fmod system :
System->init(32, FMOD_INIT_NORMAL, 0);
or you forgot to call
System->Update();
I am trying to do a certain action based on whether or not the user makes a loud sound. I'm not trying to do any voice recognition or anything. Just simply do an action based on whether the iPhone picks up a loud sound.
Any suggestions, tutorials, I can't find anything on the apple developer site. I'm assuming i'm not looking or searching right.
The easiest thing for you do is to use the AudioQueue services. Here's the manual:
Apple AQ manual
Basically, look for any example code that initialized things with AudioQueueNewInput(). Something like this:
Status = AudioQueueNewInput(&_Description,
Audio_Input_Buffer_Ready,
self,
NULL,
NULL,
0,
&self->Queue);
Once you have that going, you can enable sound level metering with something like this:
// Turn on level metering (iOS 2.0 and later)
UInt32 on = 1;
AudioQueueSetProperty(self->Queue,kAudioQueueProperty_EnableLevelMetering,&on,sizeof(on));
You will have a callback routine that is invoked for each chunk of audio data. In it, you can check the current meter levels with something like this:
//
// Check metering levels and detect silence
//
AudioQueueLevelMeterState meters[1];
UInt32 dlen = sizeof(meters);
Status = AudioQueueGetProperty(_Queue,kAudioQueueProperty_CurrentLevelMeterDB,meters,&dlen);
if (Status == 0) {
if (meters[0].mPeakPower > _threshold) {
silence = 0.0; // reset silence timer
} else {
silence += time;
}
}
//
// Notify observers of incoming data.
//
if (delegate) {
[delegate audioMeter:meters[0].mPeakPower duration:time];
[delegate audioData:Buffer->mAudioData size:Buffer->mAudioDataByteSize];
}
Or, in your case, instead of silence you can detect if the decibel level is over a certain value for long enough. Note that the decibel values you will see will range from about -70.0 for dead silence, up to 0.0db for very loud things. On an exponential scale. You'll have to play with it to see what values work for your particular application.
Apple has examples such as Speak Here which looks to have code relating to decibels. I would check some of the meter classes for examples. I have no audio programming experience but hopefully that will get you started while someone provides you with a better answer.
I am new to iPhone programming so am hoping someone can help me out here. I have searched the web, but can only find information on count down timers.
What I am looking to do is start a count up timer when a button is pressed and then stop it when a certain value drops by, say 5, and finally display that time. I can display values on screen once I have them, but getting the time in the first place is proving difficult for me.
I apologize if this is a simple question, but I look forward to reading your responses.
Thanks in advance,
stu
NSDate will provide the current date. You can use - (NSTimeInterval)timeIntervalSinceNow
to get the time since the first call and now.
There's no difference between an up-counter and down-counter. Just change the order of your subtraction.
UpcounterElapsedTime = UpcounterCurrentTime - UpcounterStartTime;
DowncounterElapsedTime = DownCounterStartTime - DownCounterCurrentTime;