AudioQueue吃了我的緩衝區（前15毫秒）

Question

我以編程方式生成音頻。我聽到我的緩衝區之間的沉默的差距。當我將手機掛在示波器上時，我發現每個緩衝區的前幾個樣本都丟失了，而且它們的位置是靜音。這種沉默的長度從幾乎沒有變化到20毫秒。

我的第一個想法是，我原來的回調函數需要太多的時間。我用盡可能最短的一個替換它 - 它一遍又一遍地重新排列相同的緩衝區。我觀察到同樣的行爲。

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的音頻API，但它似乎是類似於其他的地方通常當系統處理完第一個緩衝區，你會排隊超過一個緩衝區，這樣一來，它可以立即開始處理下一個緩衝區（因爲它已經排隊），而第一個緩衝區的完成回調正在執行。

喜歡的東西：

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); 
}