Ejemplo de uso de los servicios de cola de audio

Question

Estoy buscando un ejemplo de uso de los servicios de cola de audio.

Me gustaría crear un sonido usando una ecuación matemática y luego escucharlo.

Answer 1

Aquí está mi código para generar sonido a partir de una función. Supongo que sabe cómo usar los servicios de AudioQueue, configurar una AudioSession y comenzar y detener correctamente una cola de salida de audio.

He aquí un fragmento de cómo configurar y comenzar un AudioQueue de salida:

// 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 fields 
AudioStreamBasicDescription 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."); 
} 

Aquí está la parte que genera el tono:

// 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); 
    } 
} 

Algo a tener en cuenta es que su función se le llama en un hilo no principal, por lo que debe practicar la seguridad del hilo con bloqueos, mutex u otras técnicas.

Answer 2

alto nivel: utilizar AVAudioPlayer https://github.com/hollance/AVBufferPlayer

nivel Med: colas de audio trailsinthesand.com/exploring-iphone-audio-part-1/ consigue que va muy bien. NOTA: eliminé el http para que el enlace anterior pudiera estar allí, pero lo dirige a un sitio incorrecto, por lo que aparentemente ha cambiado.

bajo nivel: Como alternativa, puede desplegar un nivel y hacerlo con las unidades de audio: http://cocoawithlove.com/2010/10/ios-tone-generator-introduction-to.html

Answer 3

Ésta es una versión con C# de la misma muestra de @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; 
    } 

Este es el generador de tonos:

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; 
    } 
} 

también desea que estas definiciones:

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