Question

我正致力于将音频传输到iPhone，我可以将其传递给（C ++）分析算法。当然，有很多选择：AudioQueue教程at trailsinthesand开始了。

音频回调虽然提供了AudioQueueRef，但我发现Apple的文档在这方面很薄弱。内置的方法可以写入文件，但实际上你没有在数据包内查找数据。

我需要数据。我不想在文件中写任何内容，这就是所有教程 - 甚至Apple的便利I / O对象 - 似乎都是针对的。 Apple的AVAudioRecorder（令人愤怒地）将为您提供级别并写入数据，但实际上并不允许您访问它。除非我遗漏了什么......

怎么做？在下面的代码中，inBuffer->mAudioData非常接近，但我找不到有关此“数据”的格式或访问方式的信息。

AudioQueue Callback：

void AudioInputCallback(void *inUserData,
    AudioQueueRef inAQ,
    AudioQueueBufferRef inBuffer,
    const AudioTimeStamp *inStartTime,
    UInt32 inNumberPacketDescriptions,
    const AudioStreamPacketDescription *inPacketDescs)
{
    static int count = 0;
    RecordState* recordState = (RecordState*)inUserData;    
    AudioQueueEnqueueBuffer(recordState->queue, inBuffer, 0, NULL);

    ++count;
    printf("Got buffer %d\n", count);
}

将音频写入文件的代码：

OSStatus status = AudioFileWritePackets(recordState->audioFile,
                false,
                inBuffer->mAudioDataByteSize,
                inPacketDescs,
                recordState->currentPacket,
                &inNumberPacketDescriptions,
                inBuffer->mAudioData); // THIS! This is what I want to look inside of.
if(status == 0)
{
     recordState->currentPacket += inNumberPacketDescriptions;
}

Answer 1

  // so you don't have to hunt them all down when you decide to switch to float: 
  #define AUDIO_DATA_TYPE_FORMAT SInt16

  // the actual sample-grabbing code:
  int sampleCount = inBuffer->mAudioDataBytesCapacity / sizeof(AUDIO_DATA_TYPE_FORMAT);
  AUDIO_DATA_TYPE_FORMAT *samples = (AUDIO_DATA_TYPE_FORMAT*)inBuffer->mAudioData;

然后您拥有（在这种情况下为SInt16）数组samples，您可以从samples[0]访问samples[sampleCount-1]。

Answer 2

上述解决方案对我不起作用，我本身得到了错误的示例数据。（一个endian问题）如果将来有人拿错了样本数据，我希望这可以帮助你：

- （void）feedSamplesToEngine：（UInt32）audioDataBytesCapacity audioData：（void *）audioData { int sampleCount = audioDataBytesCapacity / sizeof（SAMPLE_TYPE）;

SAMPLE_TYPE *samples = (SAMPLE_TYPE*)audioData;
//SAMPLE_TYPE *sample_le = (SAMPLE_TYPE *)malloc(sizeof(SAMPLE_TYPE)*sampleCount );//for swapping endians

std::string shorts;
double power = pow(2,10);
for(int i = 0; i < sampleCount; i++)
{
    SAMPLE_TYPE sample_le =  (0xff00 & (samples[i] << 8)) | (0x00ff & (samples[i] >> 8)) ; //Endianess issue
    char dataInterim[30];
    sprintf(dataInterim,"%f ", sample_le/power); // normalize it.
    shorts.append(dataInterim);
}

如何从iOS中的AudioQueueRef获取浮点音频数据数组？

2 个答案: