15
A
답변
25
다음은 함수에서 사운드를 생성하는 코드입니다. AudioQueue 서비스를 사용하고, AudioSession을 설정하고, 오디오 출력 대기열을 올바르게 시작하고 중지하는 방법을 알고 있다고 가정합니다. 콜백이 호출 될 것입니다 조심
// 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);
}
}
뭔가 : 톤을 생성하는 부분 여기
// 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.");
}
것 : 여기
는 출력 AudioQueue을 설정하고 시작하는 코드 조각입니다 비 메인 스레드에서는 잠금, 뮤텍스 또는 다른 기술을 사용하여 스레드 안전성을 연습해야합니다.
7
높은 수준 : 당신이 잘가는 얻는다 trailsinthesand.com/exploring-iphone-audio-part-1/ 오디오 큐 : AVAudioPlayer https://github.com/hollance/AVBufferPlayer
메드 레벨을 사용합니다. 참고 : 이전 링크가있을 수 있도록 http를 제거했지만 잘못된 사이트로 직접 연결되므로 분명히 변경되었습니다.
낮은 수준 : 양자 택일로, 당신은 수준을 드롭 다운 및 오디오 장치로 작업을 수행 할 수 있습니다 http://cocoawithlove.com/2010/10/ios-tone-generator-introduction-to.html
9
이
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;
}
이가이다 @lucius에서 동일한 샘플의 C#을 사용 버전입니다 톤 제너레이터 :
0 :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;
}
}
또한 이러한 정의하려면
코드를 공유 할 수 있습니까? 신디사이저에 –