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.
Related
I am building an app that uses microphone input to detect sounds and trigger events. I based my code on AKAmplitudeTap, but I when I ran it, I found that I was only obtaining sample data for intervals with missing sections.
The tap code looks like this (with the guts ripped out and simply keeping track of how many samples would have been processed):
open class MyTap {
// internal let bufferSize: UInt32 = 1_024 // 8-9 kSamples/sec
internal let bufferSize: UInt32 = 4096 // 39.6 kSamples/sec
// internal let bufferSize: UInt32 = 16536 // 43.3 kSamples/sec
public init(_ input: AKNode?) {
input?.avAudioNode.installTap(onBus: 0, bufferSize: bufferSize, format: nil ) { buffer, _ in
sampleCount += self.bufferSize
}
}
I initialize the tap with:
func afterLoad() {
assert(!loaded)
AKSettings.audioInputEnabled = true
do {
try AKSettings.setSession(category: .playAndRecord, with: .allowBluetoothA2DP)
} catch {
print("Could not set session category.")
}
mic = AKMicrophone()
myTap = MyTap(mic) // seriously, can it be that easy?
loaded = true
}
The original tap code was capturing samples to a buffer, but I saw that big chunks of time were missing with a buffer size of 1024. I suspected that the processing time for the sample buffer might be excessive, so...
I simplified the code to simply keep track of how many samples were being passed to the tap. In another part of the code, I simply print out sampleCount/elapsedTime and, as noted in the comments after 'bufferSize' I get different amounts of samples per second.
The sample rate converges on 43.1 KSamples/sec with a 16K buffer, and only collects about 20% of the samples with a 1K buffer. I would prefer to use the small buffer size to obtain near real-time response to detected sounds. As I've been writing this, the 4K buffer version has been running and has stabilized at 39678 samples/sec.
Am I missing something? Can a tap with a small buffer size actually capture 44.1 Khz sample data?
Problem resolved... the tap requires this line of code
buffer.frameLength = self.bufferSize
... and suddenly all the samples appear. I obviously stripped out a bit too much code from the code I obviously didn't understand.
I'm using libogg and libogg, I've succeeded to add those libraries to my iPhone xCode project and encode my voice with Speex. The problem is that I cannot figure out how to pack those audio packet with ogg. Does someone know how a packet of that kind should look like or have a reference code I can use.
I know in Java it's pretty simple (you have a dedicated function for that) but not on iOS. Please help.
UPD 10.09.2013: Please, see the demo project, which basically takes pcm audiodata from wave container, encodes it with speex codec and pack everything into ogg container. Maybe later I'll create a full-fledged library/framework for all that speex routines on IOS.
UPD 16.02.2015: The demo project is republished on GitHub.
I also have been experimenting with Speex on iOS recently, with varied success, but here is something I found. Basically, if you want to pack some speex-encoded voice into an ogg file, you need to do follow three steps (assuming libogg and libspeex are already compiled and added to the project).
1) Add the first ogg page with Speex header; libspeex provides built-in tools for that (the code below is from my project, not optimal, just for the sake of example):
// create speex header
SpeexHeader spxHeader;
SpeexMode spxMode = speex_wb_mode;
int spxRate = 16000;
int spxNumberOfChannels = 1;
speex_init_header(&spxHeader, spxRate, spxNumberOfChannels, &spxMode);
// set audio and ogg packing parameters
spxHeader.vbr = 0;
spxHeader.bitrate = 16;
spxHeader.frame_size = 320;
spxHeader.frames_per_packet = 1;
// wrap speex header in ogg packet
int oggPacketSize;
_oggPacket.packet = (unsigned char *)speex_header_to_packet(&spxHeader, &oggPacketSize);
_oggPacket.bytes = oggPacketSize;
_oggPacket.b_o_s = 1;
_oggPacket.e_o_s = 0;
_oggPacket.granulepos = 0;
_oggPacket.packetno = 0;
// submit the packet to the ogg streaming layer
ogg_stream_packetin(&_oggStreamState, &_oggPacket);
free(_oggPacket.packet);
// form an ogg page
ogg_stream_flush(&_oggStreamState, &_oggPage);
// write the page to file
[_oggFile appendBytes:&_oggStreamState.header length:_oggStreamState.header_fill];
[_oggFile appendBytes:_oggStreamState.body_data length:_oggStreamState.body_fill];
2) Add the second ogg page with Vorbis comment:
// form any comment you like (I use custom struct with all fields)
vorbisCommentStruct *vorbisComment = calloc(sizeof(vorbisCommentStruct), sizeof(char));
...
// wrap Vorbis comment in ogg packet
_oggPacket.packet = (unsigned char *)vorbisComment;
_oggPacket.bytes = vorbisCommentLength;
_oggPacket.b_o_s = 0;
_oggPacket.e_o_s = 0;
_oggPacket.granulepos = 0;
_oggPacket.packetno = _oggStreamState.packetno;
// the rest should be same as in previous step
...
3) Add subsequent ogg pages with your speex-encoded audio in the similar manner.
First of all decide how many frames with audio data you want to have on every ogg page (0-255; I choose 79 quite arbitrarily):
_framesPerOggPage = 79;
Then for each frame:
// calculate current granule position of audio data within ogg file
int curGranulePos = _spxSamplesPerFrame * _oggTotalFramesCount;
// wrap audio data in ogg packet
oggPacket.packet = (unsigned char *)spxFrame;
oggPacket.bytes = spxFrameLength;
oggPacket.granulepos = curGranulePos;
oggPacket.packetno = _oggStreamState.packetno;
oggPacket.b_o_s = 0;
oggPacket.e_o_s = 0;
// submit packets to streaming layer until their number reaches _framesPerOggPage
...
// if we've reached this limit, we're ready to create another ogg page
ogg_stream_flush(&_oggStreamState, &_oggPage);
[_oggFile appendBytes:&_oggStreamState.header length:_oggStreamState.header_fill];
[_oggFile appendBytes:_oggStreamState.body_data length:_oggStreamState.body_fill];
// finally, if this is the last frame, flush all remaining packets,
// which have been created but not packed into a page, to the last page
// (don't forget to set oggPacket.e_o_s to 1 for this frame)
That's it. Hope it will help. Any corrections or questions are welcome.
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.
You must had gone through this before coming to my this question.How to use kAudioUnitSubType_LowShelfFilter of kAudioUnitType_Effect which controls bass in core Audio? Slowly & Steadily getting the things right for bass control of music. But yet not got succeeded in my objective. Now i got to know that i have to change the kAULowShelfParam_CutoffFrequency to change the bass.
The following code i was using before 5 to 7 days. this code plays music properly but doesn't change bass properly. have a look on this code snippet:-
- (void)awakeFromNib
{
printf("AUGraphController awakeFromNib\n");
mIsPlaying = false;
// clear the mSoundBuffer struct
memset(&mUserData.soundBuffer, 0, sizeof(mUserData.soundBuffer));
// create the URLs we'll use for source A and B
NSString *sourceA = [[NSBundle mainBundle] pathForResource:#"04 - Second Hand Jawaani - [rKmania.com]" ofType:#"mp3"];
NSString *sourceB = [[NSBundle mainBundle] pathForResource:#"Track2" ofType:#"mp4"];
sourceURL[0] = CFURLCreateWithFileSystemPath(kCFAllocatorDefault, (CFStringRef)sourceA, kCFURLPOSIXPathStyle, false);
sourceURL[1] = CFURLCreateWithFileSystemPath(kCFAllocatorDefault, (CFStringRef)sourceB, kCFURLPOSIXPathStyle, false);
[bassSlider addTarget:self action:#selector(selectEQPreset) forControlEvents:UIControlEventValueChanged];
}
// output unit
CAComponentDescription output_desc(kAudioUnitType_Output, kAudioUnitSubType_RemoteIO, kAudioUnitManufacturer_Apple);
// iPodEQ unit
CAComponentDescription eq_desc(kAudioUnitType_Effect, kAudioUnitSubType_AUiPodEQ, kAudioUnitManufacturer_Apple);
// multichannel mixer unit
CAComponentDescription mixer_desc(kAudioUnitType_Mixer, kAudioUnitSubType_MultiChannelMixer, kAudioUnitManufacturer_Apple);
printf("add nodes\n");
- (void)selectEQPreset;
{
AUPreset *aPreset = (AUPreset*)CFArrayGetValueAtIndex(mEQPresetsArray, value);
OSStatus result = AudioUnitSetProperty(mEQ, kAudioUnitProperty_PresentPreset, kAudioUnitScope_Global, 0, aPreset, sizeof(AUPreset));
if (result) { printf("AudioUnitSetProperty result %ld %08X %4.4s\n", result, (unsigned int)result, (char*)&result); return; };
printf("SET EQ PRESET %d ", value);
CFShow(aPreset->presetName);
}
Now after this code i got to know that i have to change frequency to change the bass, then i use the following code snippet but in the following code snippet music is not playing and its giving excess_bad too. i'm just writing the changes i made to the code.have a look on current code snippet:-
// iPodEQ unit
CAComponentDescription eq_desc(kAudioUnitType_Effect, kAudioUnitSubType_LowShelfFilter, kAudioUnitManufacturer_Apple);
- (void)selectEQPreset;
{
AudioUnit lowShelfAU;
assert(lowShelfAU);
float frequencyInHz = 120.0f;
frequencyInHz = bassSlider.value;
OSStatus result = AudioUnitSetParameter(lowShelfAU,kAULowShelfParam_CutoffFrequency,kAudioUnitScope_Global,0,frequencyInHz,0);
if (noErr != result)
{
assert(0 && "error!");
return ;
}
}
This code now i'm using but this is not changing the frequency. Its even stop playing the music and giving the excess_bad error on this line of code..
AudioUnitSetParameter(lowShelfAU,kAULowShelfParam_CutoffFrequency,kAudioUnitScope_Global,0,frequencyInHz,0);
Please anybody help me regarding this tell me how can i change the kAULowShelfParam_CutoffFrequency so that i can adjust the bass of music Via slider. Any help regarding this would be highly appreciable.
Thanks :)
Before you can use an AudioUnit you need to create it. If you're using an AUGraph your code will look something like:
AudioComponentDescription filterDescription = { kAudioUnitType_Effect, kAudioUnitSubType_LowShelfFilter, kAudioUnitSubType_LowShelfFilter, 0, 0 };
AUNode filterNode = -1;
OSStatus result = AUGraphAddNode(mAUGraph, &filterDescription, &filterNode);
if(noErr != result) {
// Handle error
}
AudioUnit filterUnit = nullptr;
result = AUGraphNodeInfo(mAUGraph, filterNode, nullptr, &filterUnit);
if(noErr != result) {
// Handle error
}
// Set parameters on filterUnit
The reason your code is failing is that the line
AudioUnit lowShelfAU;
initializes lowShelfAU with an undetermined value. An AudioUnit is a pointer type so without initialization it points to an unknown area of memory. I think it is a programming best practice to always initialize your variables when they are declared, to catch these kinds of bugs:
AudioUnit lowShelfAU = nullptr;
`AudioUnit lowShelfAU;` <-- that is an uninitialized garbage value
you need to actually create an AU instance (the Low Shelf) and add it to an AUGraph.
Note: The compiler/analyzer will identify this problem for you. I recommend people turn the warning levels waaaay up, then build, analyze, and remove all issues. Repeat until clean.
`as you know ,apple create 3 audio units according their purpose.
remoteIO --> for input and output;mixer --> for audio mixer ;for some audio effect is eq and Filter so,if u wanna cutoff the audio frequency,u need to create a filter unit and connect them together.
I'm using the Audio Queue Services API to play audio streamed from a server over a TCP socket connection on an iPhone. I can play the buffers that were filled from the socket connection, I just cannot seem to make my AudioQueue call my AudioQueueOutputCallback function, and I'm out of ideas.
High level design
Data is passed to the player from the socket connection, and written
immediately into circular buffers in memory.
As AudioQueueBuffers become available, data is copied from the circular buffers into the
available AudioQueueBuffer, which is immediately re-queued. (Or would be, if my callback happened)
What happens
The buffers are all filled and enqueued successfully, and I hear the audio stream clearly. For testing, I use a large number of buffers (15) and all of them play through seamlessly, but the AudioQueueOutputCallback is never called, so I never re-queue any of those buffers, despite the fact that everything seems to be working perfectly. If I don't wait for my callback, assuming it will never be called, and instead drive the enqueueing of buffers based on the data as it is written, I can play the audio stream indefinitely, reusing and re-enqueueing buffers as if they had been explicitly returned to me by the callback. It is that fact: that I can play the stream perfectly while reusing buffers as needed, that confuses me the most. Why isn't the callback being called?
Possibly Relevant Code
The format of the stream is 16 bit linear PCM, 8 kHz, Mono:
_streamDescription.mSampleRate = 8000.0f;
_streamDescription.mFormatID = kAudioFormatLinearPCM;
_streamDescription.mBytesPerPacket = 2;
_streamDescription.mFramesPerPacket = 1;
_streamDescription.mBytesPerFrame = sizeof(AudioSampleType);
_streamDescription.mChannelsPerFrame = 1;
_streamDescription.mBitsPerChannel = 8 * sizeof(AudioSampleType)
_streamDescription.mReserved = 0;
_streamDescription.mFormatFlags = (kLinearPCMFormatFlagIsBigEndian |
kLinearPCMFormatFlagIsPacked);
My prototype and implementation of the callback are as follows. Nothing fancy, and pretty much identical to every example I've seen so far:
// Prototype, declared above the class's #implementation
void AQBufferCallback(void* inUserData, AudioQueueRef inAudioQueue, AudioQueueBufferRef inAudioQueueBuffer);
// Definition at the bottom of the file.
void AQBufferCallback(void* inUserData, AudioQueueRef inAudioQueue, AudioQueueBufferRef inAudioQueueBuffer) {
printf("callback\n");
[(MyAudioPlayer *)inUserData audioQueue:inAudioQueue didAquireBufferForReuse:inAudioQueueBuffer];
}
I create the AudioQueue like this:
OSStatus status = 0;
status = AudioQueueNewOutput(&_streamDescription,
AQBufferCallback, // <-- Doesn't work...
self,
CFRunLoopGetCurrent(),
kCFRunLoopCommonModes,
0,
&_audioQueue);
if (status) {
// This is not called...
NSLog(#"Error creating new audio output queue: %#", [MyAudioPlayer stringForOSStatus:status]);
return;
}
And I enqueue buffers like this. At this point, it is known that the local buffer contains the correct amount of data for copying:
memcpy(aqBuffer->mAudioData, localBuffer, kAQBufferSize);
aqBuffer->mAudioDataByteSize = kAQBufferSize;
OSStatus status = AudioQueueEnqueueBuffer(_audioQueue, aqBuffer, 0, NULL);
if (status) {
// This is also not called.
NSLog(#"Error enqueueing buffer %#", [MyAudioPlayer stringForOSStatus:status]);
}
Please save me.
Is this executed on the main thread or a background thread? probably not good if CFRunLoopGetCurrent() returns a run loop of a thread that could disappear (thread pool etc) or is a run loop that don't care about kCFRunLoopCommonModes.
Try to change CFRunLoopGetCurrent() to CFRunLoopGetMain() or make sure AudioQueueNewOutput() and CFRunLoopGetCurrent() is executed on the main thread or a thread that you have control over and has a proper run loop.
Try changing self for (void*)self. Like this:
status = AudioQueueNewOutput(&_streamDescription,
AQBufferCallback,
(void*)self,
CFRunLoopGetCurrent(),
kCFRunLoopCommonModes,
0,
&_audioQueue);