I've an app that successfully uses the AUSamplerGraph to play soundfont sounds.
The main sound is guitar.
If a 'lower' note on the same string is played, I stop the previous note from playing through.
The default behaviour is like plucking a string and it rings until it dies out.
Sending a note off event, and the sound still tapers off.
Sending a note on with a volume of 0, and the sound still tapers off.
UInt32 onVelocity = loudness;
UInt32 noteCommand = kMIDIMessage_NoteOn << 4 | 0;
OSStatus osStatus;
osStatus = MusicDeviceMIDIEvent(self.samplerUnitGuitar, noteCommand, noteNbr, onVelocity, 0);
Tried quite a few things. The mode approach didn't appear to work. doesn't look like the AUSampler implements that control.
AudioUnitReset // did the trick.
To clarify, you want to stop the note's decay after you send Note Off?
To my knowledge, the only way to do this is to send a Control/Mode Change message with All Sound Off. Unfortunately, this will kill ANY notes that are currently playing on that channel. (And MusicDevice AudioUnits only respond to one channel.) But it works.
UInt8 kMIDIMessage_ControlModeChange = 0xB0;
UInt8 kMIDIMessage_ControlTypeAllSoundOff = 0x78;
MusicDeviceMIDIEvent(sampler, kMIDIMessage_ControlModeChange, kMIDIMessage_ControlTypeAllSoundOff, 0, 0);
Related
CoreAudio is always a mystery due to lack of documentations. Recently I hit some stone again:
In my program, I invoke RemoteIO and VoiceProcessingIO (VPIO) back and forth, and also change AVAudiosession in between. I tried to turn off AGC on VPIO with the follwing code:
if (ASBD.componentSubType == kAudioUnitSubType_VoiceProcessingIO) {
UInt32 turnOff = 0;
status = AudioUnitSetProperty(_myAudioUnit,
kAUVoiceIOProperty_VoiceProcessingEnableAGC,
kAudioUnitScope_Global,
0,
&turnOff,
sizeof(turnOff));
NSAssert1(status == noErr, #"Error setting AGC status: %d", (int)status);
}
Well I'm still not sure if this code disables AGC on the microphone side or the speaker side on VPIO, but anyways, let's continue. Here's the sequence to reproduce the problem:
Create a RemoteIO output audio unit with PlayAndRecord audio session category, work with it and destroy the unit;
Switch audio session to Playback only category;
Switch audio session to PlayAndRecord again and create another VPIO, work with it and destroy it;
Switch audio session to Playback and then PlayAndRecord category;
After these steps, then whatever RemoteIO/VPIO created later will bear this amplified microphone signal (as if a huge AGC is always applied) and there's no way to go back until manually kill the app and start over.
Maybe it's my particular sequence that triggered this, wonder if anyone seen this before and maybe know a correct workaround?
Try setting the mode AVAudioSessionModeMeasurement, or AVAudioSession.Mode .measurement, when configuring your app's Audio Session.
The problem occurs when I often stop and start audio playback and seek a lot back and forth in an AAC audio file through an ExtAudioFileRef object. In few cases, this strange behaviour is shown by ExtAudioFileRead:
Sometimes it assigns these numbers to the mDataByteSize of the only AudioBuffer of the AudioBufferList:
-51604480
-51227648
-51350528
-51440640
-51240960
In hex, these numbers have the pattern 0xFC....00.
The code:
status = ExtAudioFileRead(_file, &numberFramesRead, ioData);
printf("s=%li d=%p d.nb=%li, d.b.d=%p, d.b.dbs=%li, d.b.nc=%li\n", status, ioData, ioData->mNumberBuffers, ioData->mBuffers[0].mData, ioData->mBuffers[0].mDataByteSize, ioData->mBuffers[0].mNumberChannels);
Output:
s=0 d=0x16668bd0 d.nb=1, d.b.d=0x30de000, d.b.dbs=1024, d.b.nc=2 // good (usual)
s=0 d=0x16668bd0 d.nb=1, d.b.d=0x30de000, d.b.dbs=-51240960, d.b.nc=2 // misbehaving
The problem occurs on an iPhone 4S on iOS 7. I could not reproduce the problem in the Simulator.
The problem occurs when concurrently calling ExtAudioFileRead() and ExtAudioFileSeek() for the same ExtAudioFileRef from two different threads/queues.
The read function was called directly from the AURenderCallback, so it was executed on AudioUnit's real-time thread while the seek was done on my own serial queue.
I've modified the code of the render callback to also dispatch_sync() to the same serial queue to which the seek gets dispatched. That solved the problem.
I have been charged to add VOIP into an game (cross-platform, so can't use the Apple gamekit to do it).
For 3 or 4 days now, i'm trying to get my head wrap around audio unit and remoteIO...
I have overlooked tens of examples and such, but every time it is only applying a simple algorithm to the input PCM and play it back on the speaker.
According to Apple's documentation in order to do VOIP we should use kAudioSessionCategory_PlayAndRecord.
UInt32 audioCategory = kAudioSessionCategory_PlayAndRecord;
status = AudioSessionSetProperty(kAudioSessionProperty_AudioCategory,
sizeof(audioCategory),
&audioCategory);
XThrowIfError(status, "couldn't set audio category");
1) But it seems (to me) that playAndRecord will always play what coming from the mic (or more excatly the PerformThru callback // aurioTouch), am I wrong ?
I have the simplest callback, doing nothing but AURender
static OSStatus PerformThru(
void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData)
{
OSStatus err = AudioUnitRender(THIS->rioUnit, ioActionFlags, inTimeStamp, 1, inNumberFrames, ioData);
if (err)
printf("PerformThru: error %d\n", (int)err);
return err
}
From that callback I'm intending to send data to the peer (Not directly of course, but data will come from it)...
I do not see how I can play different output than the input, except maybe with 2 units, one recording, one playing, but it doesn't seems to be what Apple intended to (still accroding to the documentation).
And of course, I cannot find any documentation about it, audio unit is still pretty much un-documented...
Anyone would have an idea on what would be the best way to do it ?
I have not used VOIP or kAudioSessionCategory_PlayAndRecord. But if you want to record/transmit voice picked up from the mic and play back incoming data from network packages: Here is a good sample which included both mic and playback. Also if you have not read this doc from Apple, I would strongly recommend this.
In short: You need to create an AudioUnits instance. In it, configure two callbacks: one for mic and one for playback. The callback mic function will supply you the data that was picked up from the mic. You then can convert and transmit to other devices with whatever chosen network protocol. The playback callback function is where you supply the incoming data from other network devices to play back.
You can see this simple example. It describes how to use remote IO unit. After understanding this example, you should watch PJSIP's audio driver. These should help you implementing your own solution. Best of luck.
I'm using Audio Queue to record audio from the iphone's mic and stop recording when silence detected (no audio input for 10seconds) but I want to discard the silence from audio file.
In AudioInputCallback function I am using following code to detect silence :
AudioQueueLevelMeterState meters[1];
UInt32 dlen = sizeof(meters);
OSStatus Status AudioQueueGetProperty(inAQ,kAudioQueueProperty_CurrentLevelMeterDB,meters,&dlen);
if(meters[0].mPeakPower < _threshold)
{ // NSLog(#"Silence detected");}
But how to remove these packets? Or Is there any better option?
Instead of removing the packets from the AudioQueue, you can delay the write up by writing it to a buffer first. The buffer can be easily defined by having it inside the inUserData.
When you finish recording, if the last 10 seconds is not silent, you write it back to whatever file you are going to write. Otherwise just free the buffer.
after the file is recorded and closed, simply open and truncate the sample data you are not interested in (note: you can use AudioFile/ExtAudioFile APIs to properly update any dependent chunk/header sizes).
This captures microphone sound and changes the alpha of 'foo' according to the sound level. However, I hear the microphones input. I want the visuals to work without hearing any sound. How would I do that?
m = Microphone.get();
_root.attachAudio(m);
m.setVolume(0); //i can still hear sound. does not mute mic.
onEnterFrame = function () {
foo._alpha = m.activityLevel+33;
};
EDIT: ANSWER / SOLUTION
series8217's trick with setLoopBack did not work, but that led me to the answer online:
m = Microphone.get();
var myAudio:Sound=new Sound(attachAudio(m));
myAudio.setVolume(0);
thanks for your time
EDIT: OTHER SOLUTION
my trick may interfere with sound. using this, mutes the mic but flash still receives input:
m = Microphone.get();
m.setSilenceLevel(100);
Switching the loopback mode on the microphone object should do the trick.
m.setLoopBack(false);
However, if that doesn't do it, perhaps your OS sound settings have monitor or loopback mode turned on. I'd say look into that before trying setLoopback().