Example of using Audio Queue Services

Here's my code for generating sound from a function. I'm assuming you know how to use AudioQueue services, set up an AudioSession, and properly start and stop an audio output queue.

Here's a snippet for setting up and starting an output AudioQueue:

// Get the preferred sample rate (8,000 Hz on iPhone, 44,100 Hz on iPod touch)size = sizeof(sampleRate);err = AudioSessionGetProperty (kAudioSessionProperty_CurrentHardwareSampleRate, &size, &sampleRate);if (err != noErr) NSLog(@"AudioSessionGetProperty(kAudioSessionProperty_CurrentHardwareSampleRate) error: %d", err); //NSLog (@"Current hardware sample rate: %1.0f", sampleRate);BOOL isHighSampleRate = (sampleRate > 16000);int bufferByteSize;AudioQueueBufferRef buffer;// Set up stream format fieldsAudioStreamBasicDescription streamFormat;streamFormat.mSampleRate = sampleRate;streamFormat.mFormatID = kAudioFormatLinearPCM;streamFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kLinearPCMFormatFlagIsPacked;streamFormat.mBitsPerChannel = 16;streamFormat.mChannelsPerFrame = 1;streamFormat.mBytesPerPacket = 2 * streamFormat.mChannelsPerFrame;streamFormat.mBytesPerFrame = 2 * streamFormat.mChannelsPerFrame;streamFormat.mFramesPerPacket = 1;streamFormat.mReserved = 0;// New output queue ---- PLAYBACK ----if (isPlaying == NO) {    err = AudioQueueNewOutput (&streamFormat, AudioEngineOutputBufferCallback, self, nil, nil, 0, &outputQueue);    if (err != noErr) NSLog(@"AudioQueueNewOutput() error: %d", err);    // Enqueue buffers    //outputFrequency = 0.0;    outputBuffersToRewrite = 3;    bufferByteSize = (sampleRate > 16000)? 2176 : 512; // 40.5 Hz : 31.25 Hz     for (i=0; i<3; i++) {        err = AudioQueueAllocateBuffer (outputQueue, bufferByteSize, &buffer);         if (err == noErr) {            [self generateTone: buffer];            err = AudioQueueEnqueueBuffer (outputQueue, buffer, 0, nil);            if (err != noErr) NSLog(@"AudioQueueEnqueueBuffer() error: %d", err);        } else {            NSLog(@"AudioQueueAllocateBuffer() error: %d", err);             return;        }    }    // Start playback    isPlaying = YES;    err = AudioQueueStart(outputQueue, nil);    if (err != noErr) { NSLog(@"AudioQueueStart() error: %d", err); isPlaying= NO; return; }} else {    NSLog (@"Error: audio is already playing back.");}

Here's the part that generates the tone:

// AudioQueue output queue callback.void AudioEngineOutputBufferCallback (void *inUserData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) {    AudioEngine *engine = (AudioEngine*) inUserData;    [engine processOutputBuffer:inBuffer queue:inAQ];}- (void) processOutputBuffer: (AudioQueueBufferRef) buffer queue:(AudioQueueRef) queue {    OSStatus err;    if (isPlaying == YES) {        [outputLock lock];        if (outputBuffersToRewrite > 0) {            outputBuffersToRewrite--;            [self generateTone:buffer];        }        err = AudioQueueEnqueueBuffer(queue, buffer, 0, NULL);        if (err == 560030580) { // Queue is not active due to Music being started or other reasons            isPlaying = NO;        } else if (err != noErr) {            NSLog(@"AudioQueueEnqueueBuffer() error %d", err);        }        [outputLock unlock];    } else {        err = AudioQueueStop (queue, NO);        if (err != noErr) NSLog(@"AudioQueueStop() error: %d", err);    }}-(void) generateTone: (AudioQueueBufferRef) buffer {    if (outputFrequency == 0.0) {        memset(buffer->mAudioData, 0, buffer->mAudioDataBytesCapacity);        buffer->mAudioDataByteSize = buffer->mAudioDataBytesCapacity;    } else {        // Make the buffer length a multiple of the wavelength for the output frequency.        int sampleCount = buffer->mAudioDataBytesCapacity / sizeof (SInt16);        double bufferLength = sampleCount;        double wavelength = sampleRate / outputFrequency;        double repetitions = floor (bufferLength / wavelength);        if (repetitions > 0.0) {            sampleCount = round (wavelength * repetitions);        }        double      x, y;        double      sd = 1.0 / sampleRate;        double      amp = 0.9;        double      max16bit = SHRT_MAX;        int i;        SInt16 *p = buffer->mAudioData;        for (i = 0; i < sampleCount; i++) {            x = i * sd * outputFrequency;            switch (outputWaveform) {                case kSine:                     y = sin (x * 2.0 * M_PI);                    break;                case kTriangle:                    x = fmod (x, 1.0);                    if (x < 0.25)                        y = x * 4.0; // up 0.0 to 1.0                    else if (x < 0.75)                        y = (1.0 - x) * 4.0 - 2.0; // down 1.0 to -1.0                    else                         y = (x - 1.0) * 4.0; // up -1.0 to 0.0                    break;                case kSawtooth:                    y  = 0.8 - fmod (x, 1.0) * 1.8;                    break;                case kSquare:                    y = (fmod(x, 1.0) < 0.5)? 0.7: -0.7;                    break;                default: y = 0; break;            }            p[i] = y * max16bit * amp;        }        buffer->mAudioDataByteSize = sampleCount * sizeof (SInt16);    }}

Something to watch out for is that your callback will be called on a non-main thread, so you have to practice thread safety with locks, mutexs, or other techniques.

This is a version using C# of the same sample from @lucius

    void SetupAudio ()    {        AudioSession.Initialize ();        AudioSession.Category = AudioSessionCategory.MediaPlayback;        sampleRate = AudioSession.CurrentHardwareSampleRate;        var format = new AudioStreamBasicDescription () {            SampleRate = sampleRate,            Format = AudioFormatType.LinearPCM,            FormatFlags = AudioFormatFlags.LinearPCMIsSignedInteger | AudioFormatFlags.LinearPCMIsPacked,            BitsPerChannel = 16,            ChannelsPerFrame = 1,            BytesPerFrame = 2,            BytesPerPacket = 2,             FramesPerPacket = 1,        };        var queue = new OutputAudioQueue (format);        var bufferByteSize = (sampleRate > 16000)? 2176 : 512; // 40.5 Hz : 31.25 Hz         var buffers = new AudioQueueBuffer* [numBuffers];        for (int i = 0; i < numBuffers; i++){            queue.AllocateBuffer (bufferByteSize, out buffers [i]);            GenerateTone (buffers [i]);            queue.EnqueueBuffer (buffers [i], null);        }        queue.OutputCompleted += (object sender, OutputCompletedEventArgs e) => {            queue.EnqueueBuffer (e.UnsafeBuffer, null);        };        queue.Start ();        return true;    }

This is the tone generator:

    void GenerateTone (AudioQueueBuffer *buffer)    {        // Make the buffer length a multiple of the wavelength for the output frequency.        uint sampleCount = buffer->AudioDataBytesCapacity / 2;        double bufferLength = sampleCount;        double wavelength = sampleRate / outputFrequency;        double repetitions = Math.Floor (bufferLength / wavelength);        if (repetitions > 0)             sampleCount = (uint)Math.Round (wavelength * repetitions);        double      x, y;        double      sd = 1.0 / sampleRate;        double      amp = 0.9;        double      max16bit = Int16.MaxValue;        int i;        short *p = (short *) buffer->AudioData;        for (i = 0; i < sampleCount; i++) {            x = i * sd * outputFrequency;            switch (outputWaveForm) {                case WaveForm.Sine:                     y = Math.Sin (x * 2.0 * Math.PI);                    break;                case WaveForm.Triangle:                    x = x % 1.0;                    if (x < 0.25)                        y = x * 4.0; // up 0.0 to 1.0                    else if (x < 0.75)                        y = (1.0 - x) * 4.0 - 2.0; // down 1.0 to -1.0                    else                         y = (x - 1.0) * 4.0; // up -1.0 to 0.0                    break;                case WaveForm.Sawtooth:                    y  = 0.8 - (x % 1.0) * 1.8;                    break;                case WaveForm.Square:                    y = ((x % 1.0) < 0.5)? 0.7: -0.7;                    break;                default: y = 0; break;            }            p[i] = (short)(y * max16bit * amp);        }        buffer->AudioDataByteSize = sampleCount * 2;    }}

You also want these definitions:

    enum WaveForm {        Sine, Triangle, Sawtooth, Square    }    WaveForm outputWaveForm;    const float outputFrequency = 220;

High level: use AVAudioPlayer https://github.com/hollance/AVBufferPlayer

Med level: audio queues trailsinthesand.com/exploring-iphone-audio-part-1/ gets you going nicely. NOTE: I removed the http so the old link could be there, but it does direct to a bad site, so it apparently has changed.

Low level: alternatively, you can drop down a level and do it with audio units: http://cocoawithlove.com/2010/10/ios-tone-generator-introduction-to.html