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间隙。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，取而代之的是静默。这种静默的长度从几乎没有变化到长达 20 毫秒。

我的第一个想法是我原来的回调函数花费了太多时间。我用最短的可能替换它——它一遍又一遍地重新排列相同的缓冲区。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，但它似乎类似于其他通常您会排队多个缓冲区的 API，这样当系统完成处理第一个缓冲区时，它可以立即开始处理正在执行第一个缓冲区的完成回调时，下一个缓冲区（因为它已经排队）。

类似：

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);
}