これが関数から音を生成するための私のコードです。 AudioQueueサービスの使用方法、AudioSessionの設定方法、およびオーディオ出力キューを適切に開始および停止する方法を知っていることを前提としています。
出力AudioQueueを設定および開始するためのスニペットは次のとおりです。
// 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.");
}
トーンを生成する部分は次のとおりです。
// 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);
}
}
注意すべき点は、コールバックが非メインスレッドで呼び出されることです。そのため、ロック、ミューテックス、またはその他の手法を使用してスレッドセーフを練習する必要があります。
これは、@ luciusの同じサンプルのC＃を使用したバージョンです。
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;
}
これはトーンジェネレータです：
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;
}
}
次の定義も必要です。
enum WaveForm {
Sine, Triangle, Sawtooth, Square
}
WaveForm outputWaveForm;
const float outputFrequency = 220;
高レベル：AVAudioPlayerを使用 https://github.com/hollance/AVBufferPlayer
中程度のレベル：オーディオキューtrailsinthesand.com/exploring-iphone-audio-part-1/でうまくいきます。注：古いリンクが存在する可能性があるため、httpを削除しましたが、悪いサイトに直接リンクしているため、明らかに変更されています。
低レベル：または、レベルを下げてオーディオユニットで実行することもできます： http://cocoawithlove.com/2010/10/ios-tone-generator-introduction-to.html