objective-c audio core-audio audioqueueservices

objective c - Ejemplo de uso de servicios de cola de audio



objective-c core-audio (3)

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

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

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

Aquí hay un fragmento para configurar e iniciar una salida de 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 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 devolución de llamada se llamará en un subproceso no principal, por lo que tiene que practicar la seguridad de subprocesos con bloqueos, exclusión mutua u otras técnicas.

Esta es una versión que usa 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 quieres estas definiciones:

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