I used an AVAudioPlayer object to control playing multiple music files. I also created an UISlider to control seeking file. But i have a problem when seek the pointer. After seeking, AVAudioPlayer update time correct then jump into "audioPlayerDidFinishPlaying" function unexpected.
Here is the code that i used :
-(void)timeChange
{
_player.currentTime = _timeControl.value;
[self updateCurrentTimeForPlayer];
}
-(void)updateCurrentTimeForPlayer
{
if(_isNeedUpdate == NO) return;
_timeControl.maximumValue = _player.duration;
}
A long shot, but maybe the audio format doesn't support seeking?
Why the call to updateCurrentTimeForPlayer? Where is _isNeedUpdate set? (Why all the underscores?)
Can you add some debug NSLogs to find out what's going on?
Related
Ive got the following code to check for agent collision.
I want to fire a MIDI message only once when they start colliding.
Ive got this so far.
void draw(){
//Loop through people, and check collision, then play note, if intersecting
for(int i=0;i<people.size();i++){
Person p = people.get(i);
p.collide(people,collisions);
p.triggerMidi();
p.run();
}
}
public void collide(ArrayList<Person> people, ArrayList<Person> connections) {
for(Person other : people) {
if (other != this) {
if (this.collide(other)) {
this.isIntersecting=true;
//connections.add(other); // when a collision is found, add it to a list for later use.
}
}
}
}
void triggerMidi(){
if(!hasPlayed && this.isIntersecting==true){
MIDI.sendNoteOn(channel, agentNote, 127);
delay(200);
MIDI.sendNoteOff(channel,agentNote, 127);
hasPlayed=true;
}
}
This works to play the sound only once at the start of collision.
But how do I get it to play again at the start of another collision.
Obviously I have to set hasPlayed back to false.
But where?
When I set it to false in the collide loop, the sound play a million times.
Any ideas?
First off, you probably shouldn't have a call to delay() from your drawing thread. That will cause your sketch to become laggy and unresponsive. Instead, you might want to put your sound playing on a different thread.
Then, to answer your original question- do you know how long the note plays for? If so, just record the time that the note starts, and then use that time to check the elapsed time. The millis() function might come in handy for that. When the elapsed time is greater than the duration of the note, then you can set hasPlayed back to false.
I'm generating tones on iPhone using AudioUnits based on Matt Gallagher's classic example. In order to avoid the chirps and clicks at the beginning/end, I'm fading the amplitude in/out in the RenderTone callback. I'd like to destroy the ToneUnit at the end of the fade out, that is, after the amplitude reaches zero. The only way I can think to do this is to call an instance method from within the callback:
if (PlayerState == FADING_OUT) {
amplitude -= stepsize;
if (amplitude <= 0) {
amplitude = 0;
PlayerState = OFF;
[viewController destroyToneUnit];
}
}
Unfortunately this is more challenging that I had thought. For one thing, I still get the click at the end that the fadeout was supposed to eliminate. For another, I get this log notice:
<AURemoteIO::IOThread> Someone is deleting an AudioConverter while it is in use.
What does this message mean and why am I getting it?
How should I kill the ToneUnit? I suspect that the click occurs because RenderTone and destroyToneUnit run on different threads. How can I get these synchronized?
In case it's helpful, here's my destroyToneUnit instance method:
- (void) destroyToneUnit {
AudioOutputUnitStop(toneUnit);
AudioUnitUninitialize(toneUnit);
AudioComponentInstanceDispose(toneUnit);
toneUnit = nil;
}
If I NSLog messages right before and right after AudioUnitUninitialize(toneUnit);, the notice appears between them.
I also ran into the same issue. When I called the destroyToneUnit from the main thread, the warning went away.
[viewController performSelectorOnMainThread:#selector(destroyToneUnit) withObject:nil waitUntilDone:NO];
I'm building an iPhone app that generates random guitar music by playing back individual recorded guitar notes in "caf" format. These notes vary in duration from 3 to 11 seconds, depending on the amount of sustain.
I originally used the AVAudioPlayer for playback, and in the simulator at 120 bpm, playing 16th notes it sung beautifully, but on my handset, as soon as I
upped the tempo a little over 60 bpm playing just 1/4 notes, it ran like a dog and wouldn't keep in time. My elation was very short lived.
To reduce latency, I tried to implement playback via Audio Units using the Apple MixerHost project as a template for an audio engine, but kept getting a bad access error after I bolted it on and connected everything up.
After many hours of it doing my head in, I gave up on that avenue of thought and I bolted on the Novocaine audio engine instead.
I have now run into a brick wall trying to connect it up to my model.
On the most basic level, my model is a Neck object containing an NSDictionary of Note objects.
Each Note object knows what string and fret of the guitar neck it's on and contains its own AVAudioPlayer.
I build a chromatic guitar neck containing either 122 notes (6 strings by 22 frets) or 144 notes (6 strings by 24 frets) depending on the neck size selected in the user preferences.
I use these Notes as my single point of truth so all scalar Notes generated by the music engine are pointers to this chromatic note bucket.
#interface Note : NSObject <NSCopying>
{
NSString *name;
AVAudioPlayer *soundFilePlayer;
int stringNumber;
int fretNumber;
}
I always start off playback with the root Note or Chord of the selected scale and then generate the note to play next so I am always playing one note behind the generated note. This way, the next Note to play is always queued up ready to go.
Playback control of these Notes is a achieved with the following code:
- (void)runMusicGenerator:(NSNumber *)counter
{
if (self.isRunning) {
Note *NoteToPlay;
// pulseRate is the time interval between beats
// staticNoteLength = 1/4 notes, 1/8th notes, 16th notes, etc.
float delay = self.pulseRate / [self grabStaticNoteLength];
// user setting to play single, double or triplet notes.
if (self.beatCounter == CONST_BEAT_COUNTER_INIT_VAL) {
NoteToPlay = [self.GuitarNeck generateNoteToPlayNext];
} else {
NoteToPlay = [self.GuitarNeck cloneNote:self.GuitarNeck.NoteToPlayNow];
}
self.GuitarNeck.NoteToPlayNow = NoteToPlay;
[self callOutNoteToPlay];
[self performSelector:#selector(runDrill:) withObject:NoteToPlay afterDelay:delay];
}
- (Note *)generateNoteToPlayNext
{
if ((self.musicPaused) || (self.musicStopped)) {
// grab the root note on the string to resume
self.NoteToPlayNow = [self grabRootNoteForString];
//reset the flags
self.musicPaused = NO;
self.musicStopped = NO;
} else {
// Set NoteRingingOut to NoteToPlayNow
self.NoteRingingOut = self.NoteToPlayNow;
// Set NoteToPlaNowy to NoteToPlayNext
self.NoteToPlayNow = self.NoteToPlayNext;
if (!self.NoteToPlayNow) {
self.NoteToPlayNow = [self grabRootNoteForString];
// now prep the note's audio player for playback
[self.NoteToPlayNow.soundFilePlayer prepareToPlay];
}
}
// Load NoteToPlayNext
self.NoteToPlayNext = [self generateRandomNote];
}
- (void)callOutNoteToPlay
{
self.GuitarNeck.NoteToPlayNow.soundFilePlayer.delegate = (id)self;
[self.GuitarNeck.NoteToPlayNow.soundFilePlayer setVolume:1.0];
[self.GuitarNeck.NoteToPlayNow.soundFilePlayer setCurrentTime:0];
[self.GuitarNeck.NoteToPlayNow.soundFilePlayer play];
}
Each Note's AVAudioPlayer is loaded as follows:
- (AVAudioPlayer *)buildStringNotePlayer:(NSString *)nameOfNote
{
NSString *soundFileName = #"S";
soundFileName = [soundFileName stringByAppendingString:[NSString stringWithFormat:#"%d", stringNumber]];
soundFileName = [soundFileName stringByAppendingString:#"F"];
if (fretNumber < 10) {
soundFileName = [soundFileName stringByAppendingString:#"0"];
}
soundFileName = [soundFileName stringByAppendingString:[NSString stringWithFormat:#"%d", fretNumber]];
NSString *soundPath = [[NSBundle mainBundle] pathForResource:soundFileName ofType:#"caf"];
NSURL *fileURL = [NSURL fileURLWithPath:soundPath];
AVAudioPlayer *audioPlayer = [[AVAudioPlayer alloc] initWithContentsOfURL:fileURL error:nil];
return notePlayer;
}
Here is where I come a cropper.
According to the Novocaine Github page ...
Playing Audio
Novocaine *audioManager = [Novocaine audioManager];
[audioManager setOutputBlock:^(float *audioToPlay, UInt32 numSamples, UInt32 numChannels) {
// All you have to do is put your audio into "audioToPlay".
}];
But in the downloaded project, you use the following code to load the audio ...
// AUDIO FILE READING OHHH YEAHHHH
// ========================================
NSURL *inputFileURL = [[NSBundle mainBundle] URLForResource:#"TLC" withExtension:#"mp3"];
fileReader = [[AudioFileReader alloc]
initWithAudioFileURL:inputFileURL
samplingRate:audioManager.samplingRate
numChannels:audioManager.numOutputChannels];
[fileReader play];
fileReader.currentTime = 30.0;
[audioManager setOutputBlock:^(float *data, UInt32 numFrames, UInt32 numChannels)
{
[fileReader retrieveFreshAudio:data numFrames:numFrames numChannels:numChannels];
NSLog(#"Time: %f", fileReader.currentTime);
}];
Here is where I really start to get confused because the first method uses a float and the second one uses a URL.
How do you pass a "caf" file to a float? I am not sure how to implement Novocaine - it is still fuzzy in my head.
My questions that I hope someone can help me with are as follows ...
Are Novocaine objects similar to AVAudioPlayer objects, just more versatile and tweaked to the max for minimum latency? i.e. self contained audio playing (/recording/generating) units?
Can I use Novocaine in my model as it is? i.e. 1 Novocaine object per chromatic note or should I have 1 novocain object that contains all the Chromatic Notes? Or do I just store the URL in the note instead and pass that to a Novocaine player?
How can I put my audio into "audioToPlay" when my audio is a "caf" file and "audioToPlay" take a float?
If I include and declare a Novocaine property in Note.m do I then have to rename the class to Note.mm in order to use the Novocaine object?
How do I play multiple Novocaine objects concurrently in order to reproduce chords and intervals?
Can I loop a Novocaine object's playback?
Can I set the playback length of a note? i.e. play a 10 sec note for only 1 sec?
Can I modify the above code to use Novocaine?
Is the method I am using for runMusicGenerator the correct one to use in order to maintain a tempo that is up to professional standards?
Novocaine makes your life easier by eliminating the need for you to setup the RemoteIO AudioUnit manually. This includes having to painfully fill a bunch of CoreAudio structs and providing a bunch of callbacks such as this audio process callback.
static OSStatus PerformThru(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *inTimeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData);
Instead Novocaine handles that in its implementation and then calls your block, which you set by doing this.
[audioManager setOutputBlock: ^(float *audioToPlay, UInt32 numSamples, UInt32 numChannels){} ];
Whatever you write to audioToPlay gets played.
Novocaine sets up the RemoteIO AudioUnit for you. This is a low-level CoreAudio API, different from the high-level AVFoundation, and very low-latency as expected. You are right in that Novocaine is self-contained. You can record, generate, and process audio in realtime.
Novocaine is a singleton, you cannot have multiple Novocaine instances. One way to do it is to store your guitar sound/sounds in a separate class or array, and then write a bunch of methods, using Novocaine to play them.
You have a bunch of options. You can use Novocaine's AudioFileReader to play your .caf file for you. You do this by allocating an AudioFileReader and then passing the URL of the .caf file you want to play, as per example code. You then stick [fileReader retrieveFreshAudio:data numFrames:numFrames numChannels:numChannels] in your block, as per example code. Each time your block is called, AudioFileReader grabs and buffers a chunk of audio from disk and puts it in audioToPlay which subsequently gets played. There are some disadvantages with this. For short sounds (such as your guitar sound I'm assuming) repeatedly calling retrieveFreshAudio is a performance hit. It is generally a better idea (for short sounds) to perform a synchronous, sequential read of the entire file into memory. Novocaine does not provide a way to do this (yet). You will have to use ExtAudioFileServices to do this. The Apple example project MixerHost details how to do this.
If you are using AudioFileReader yes. You only rename to .mm when you are #import ing from Obj-C++ headers or #include ing C++ headers.
As mentioned earlier, only 1 Novocaine instance is allowed. You can achieve polyphony by mixing multiple audio sources. This is simply just adding buffers together. If you have made multiple versions of the same guitar sound at different pitches, just read them all in to memory, and mix away. If you only want to have one guitar sound, then you have to, in realtime, change the playback rate of however many notes you are playing and then mixdown.
Novocaine is agnostic to what you are actually playing and does not care how long you are playing a sample for. In order to loop a sound, you have to maintain a count of how many samples have elapsed, check if you are at the end of your sound, and then set that count back to 0.
Yes. Assuming a 44.1k sample rate, 1 sec of audio = 44100 samples. You would then reset your count when it reaches 44100.
Yes. It looks something like this. Assuming you have 4 guitar sounds which are mono and longer than 1 second long, and you have read them into memory float *guitarC, *guitarE, *guitarG, *guitarB; (jazzy CMaj7 chord w00t), and want to mix them down for 1 second and loop that back in mono:
[audioManager setOutputBlock:^(float *data, UInt32 numFrames, UInt32 numChannels){
static int count = 0;
for(int i=0; i<numFrames; ++i){
//Mono mix each sample of each sound together. Since result can be 4x louder, divide the total amp by 4.
//You should be using `vDSP_vadd` from the accelerate framework for added performance.
data[count] = (guitarC[count] + guitarE[count] + guitarG[count] + guitarB[count]) * 0.25;
if(++count >= 44100) count = 0; //Plays the mix for 1 sec
}
}];
Not exactly. Using performSelector or any mechanism scheduled on a runloop or thread is not guaranteed to be precise. You might experience timing irregularities when the CPU load fluctuates, for example. Use the audio block if you want sample accurate timing.
I am currently working on an audio DSP App development. The project requires direct access and modification of audio data. Right now I can successfully access and modify the raw audio data using AudioQueue but encounters error during playback. The output audio after any modification turns out be noise.
In short, the code is something like this:
(Modified from Speakhere sample code. The rest remains unchanged.)
void AQPlayer::AQBufferCallback(void * inUserData,
AudioQueueRef inAQ,
AudioQueueBufferRef inCompleteAQBuffer)
{
AQPlayer *THIS = (AQPlayer *)inUserData;
if (THIS->mIsDone) return;
UInt32 numBytes;
UInt32 nPackets = THIS->GetNumPacketsToRead();
OSStatus result = AudioFileReadPackets(THIS->GetAudioFileID(),
false,
&numBytes,
inCompleteAQBuffer->mPacketDescriptions,
THIS->GetCurrentPacket(),
&nPackets,
inCompleteAQBuffer->mAudioData);
if (result)
printf("AudioFileReadPackets failed: %d", (int)result);
if (nPackets > 0) {
inCompleteAQBuffer->mAudioDataByteSize = numBytes;
inCompleteAQBuffer->mPacketDescriptionCount = nPackets;
//My modification starts from here
//Modifying audio data
SInt16 *testBuffer = (SInt16*)inCompleteAQBuffer->mAudioData;
for (int i = 0; i < (inCompleteAQBuffer->mAudioDataByteSize)/sizeof(SInt16); i++)
{
//printf("before modification %d", (int)*testBuffer);
*testBuffer = (SInt16) *testBuffer/2; //Say some simple modification
//printf("after modification %d", (int)*testBuffer);
testBuffer++;
}
AudioQueueEnqueueBuffer(inAQ, inCompleteAQBuffer, 0, NULL);
}
During debugging, the data in buffer is displayed as expected, but the actual output is nothing but noise.
Here are some other strange behaviors of the code that makes both the whole team crazy:
If there is no change to the data (add/sub by 0, multiply by 1) or the whole buffer is assigned to a constant (say 0, then the audio will be muted), the playback behaves normally (Of course!) But if I perform anything more than it, it still turns out to be noise.
In the case I hardcode a single tone as test audio, the output noise spreads into another channel also.
So where is the bug in this code? Or if I am on the wrong track, what is the correct approach to modify the audio data and perform playback CORRECTLY? Any insight will be sincerely appreciated.
Thank you very much :-)
Cheers,
Manca
are you SURE the sample format is SInt16? And how many channels are there? You seem to treat the audio as a single channel short stream, but suppose the format is actually dual channel Float32 or so, and you do the modifications there, than the effect would be exactly as you describe, including the noise on other channels.
In my application I'm using following coding pattern to vibrate my iPhone device
Include: AudioToolbox framework
Header File:
#import "AudioToolbox/AudioServices.h"
Code:
AudioServicesPlaySystemSound(kSystemSoundID_Vibrate);
My problem is that when I run my application it gets vibrate but only for second but I want that it will vibrate continuously until I will stop it.
How could it be possible?
Thankfully, it's not possible to change the duration of the vibration. The only way to trigger the vibration is to play the kSystemSoundID_Vibrate as you have. If you really want to though, what you can do is to repeat the vibration indefinitely, resulting in a pulsing vibration effect instead of a long continuous one. To do this, you need to register a callback function that will get called when the vibration sound that you play is complete:
AudioServicesAddSystemSoundCompletion (
kSystemSoundID_Vibrate,
NULL,
NULL,
MyAudioServicesSystemSoundCompletionProc,
NULL
);
AudioServicesPlaySystemSound(kSystemSoundID_Vibrate);
Then you define your callback function to replay the vibrate sound again:
#pragma mark AudioService callback function prototypes
void MyAudioServicesSystemSoundCompletionProc (
SystemSoundID ssID,
void *clientData
);
#pragma mark AudioService callback function implementation
// Callback that gets called after we finish buzzing, so we
// can buzz a second time.
void MyAudioServicesSystemSoundCompletionProc (
SystemSoundID ssID,
void *clientData
) {
if (iShouldKeepBuzzing) { // Your logic here...
AudioServicesPlaySystemSound(kSystemSoundID_Vibrate);
} else {
//Unregister, so we don't get called again...
AudioServicesRemoveSystemSoundCompletion(kSystemSoundID_Vibrate);
}
}
There are numerous examples that show how to do this with a private CoreTelephony call: _CTServerConnectionSetVibratorState, but it's really not a sensible course of action since your app will get rejected for abusing the vibrate feature like that. Just don't do it.
Read the Apple Human Interaction Guidelines for iPhone. I believe this is not approved behavior in an app.
iOS 5 has implemented Custom Vibrations mode. So in some cases variable vibration is acceptable. The only thing is unknown what library deals with that (pretty sure not CoreTelephony) and if it is open for developers. So keep on searching.
The above answers are good and you can do it in a simple way also.
You can use the recursive method calls.
func vibrateTheDeviceContinuously() throws {
// Added concurrent queue for next & Vibrate device
DispatchQueue.global(qos: .utility).async {
//Vibrate the device
AudioServicesPlaySystemSound(kSystemSoundID_Vibrate)
self.incrementalCount += 1
usleep(800000) // if you don't want pause in between, remove this line.
do {
if let isKeepBuzzing = self.iShouldKeepBuzzing , isKeepBuzzing == true {
try self.vibrateTheDeviceContinuously()
}
else {
return
}
} catch {
//Exception handle
print("exception")
}
}
}
To stop the device vibration use the following line.
self.iShouldKeepBuzzing = false
ios swift