AudioQueue吃了我的缓冲区(前15毫秒)

Question

我正在以编程方式生成音频.我听到缓冲区之间的沉默差距.当我将手机连接到示波器时,我发现每个缓冲区的前几个样本都丢失了,而它们的位置则是静音.这种沉默的长度从几乎没有变化到20毫秒.

我的第一个想法是我原来的回调函数需要花费太多时间.我用尽可能短的替换它 - 它反复重新排队相同的缓冲区.我观察到同样的行为.

AudioQueueRef aq;
AudioQueueBufferRef aq_buffer;
AudioStreamBasicDescription asbd;

void aq_callback (void *aqData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) {
    OSStatus s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL);
}

void aq_init(void) {
    OSStatus s;

    asbd.mSampleRate = AUDIO_SAMPLES_PER_S;
    asbd.mFormatID = kAudioFormatLinearPCM;
    asbd.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; 
    asbd.mBytesPerPacket = 1;
    asbd.mFramesPerPacket = 1; 
    asbd.mBytesPerFrame = 1;
    asbd.mChannelsPerFrame = 1;
    asbd.mBitsPerChannel = 8;
    asbd.mReserved = 0;


    int PPM_PACKETS_PER_SECOND = 50;
    // one buffer is as long as one PPM frame
    int BUFFER_SIZE_BYTES = asbd.mSampleRate/PPM_PACKETS_PER_SECOND*asbd.mBytesPerFrame;

    s = AudioQueueNewOutput(&asbd, aq_callback, NULL, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &aq);
    s = AudioQueueAllocateBuffer(aq, BUFFER_SIZE_BYTES, &aq_buffer);

    // put samples in the buffer
    buffer_data(my_data, aq_buffer);

    s = AudioQueueStart(aq, NULL);
    s = AudioQueueEnqueueBuffer(aq, aq_buffer, 0, NULL);
}

Answer 1

我不熟悉iPhone音频API,但它似乎与其他通常会排队多个缓冲区的类似,这样当系统处理完第一个缓冲区时,它可以立即开始处理下一个缓冲区(因为它已经排队)正在执行第一个缓冲区上的完成回调.

就像是:

AudioQueueRef aq;
AudioQueueBufferRef aq_buffer[2];
AudioStreamBasicDescription asbd;

void aq_callback (void *aqData, AudioQueueRef inAQ, AudioQueueBufferRef inBuffer) {
    // note that the callback tells us which buffer has been completed, so all
    //  we have to do is queue it back up
    OSStatus s = AudioQueueEnqueueBuffer(aq, inBuffer, 0, NULL);
}

void aq_init(void) {
    OSStatus s;

    asbd.mSampleRate = AUDIO_SAMPLES_PER_S;
    asbd.mFormatID = kAudioFormatLinearPCM;
    asbd.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; 
    asbd.mBytesPerPacket = 1;
    asbd.mFramesPerPacket = 1; 
    asbd.mBytesPerFrame = 1;
    asbd.mChannelsPerFrame = 1;
    asbd.mBitsPerChannel = 8;
    asbd.mReserved = 0;


    int PPM_PACKETS_PER_SECOND = 50;
    // one buffer is as long as one PPM frame
    int BUFFER_SIZE_BYTES = asbd.mSampleRate/PPM_PACKETS_PER_SECOND*asbd.mBytesPerFrame;

    s = AudioQueueNewOutput(&asbd, aq_callback, NULL, CFRunLoopGetCurrent(), kCFRunLoopCommonModes, 0, &aq);
    s = AudioQueueAllocateBuffer(aq, BUFFER_SIZE_BYTES, &aq_buffer[0]);
    s = AudioQueueAllocateBuffer(aq, BUFFER_SIZE_BYTES, &aq_buffer[1]);

    // put samples in the buffer - fill both buffers
    buffer_data(my_data, aq_buffer[0]);
    buffer_data(my_data, aq_buffer[1]);

    s = AudioQueueStart(aq, NULL);
    s = AudioQueueEnqueueBuffer(aq, aq_buffer[0], 0, NULL);
    s = AudioQueueEnqueueBuffer(aq, aq_buffer[1], 0, NULL);
}