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我正在使用CoreAudio低级音频采集API。应用程序目标是MAC OSX，而不是iOS。由于音频数据错误而导致音频破裂
在测试过程中，我们不时用真实的音频对噪声进行非常烦人的调制。这些现象随着时间而发展，从几乎没有引人注目的地步开始，变得越来越占统治地位。
分析Audacity下的捕获音频表明音频数据包的末尾部分是错误的。
下面是样品图像： 
侵入重复每40ms哪个是所配置的分组时间（在缓冲液中的样品计）
更新： 随着时间的推移的间隙变大，这里是10分钟后，来自同一捕获文件的另一个快照。现在差距包含1460个样本，距离包总数为40毫秒33毫秒！ 
CODE SNIPPESTS：
捕捉回调
OSStatus MacOS_AudioDevice::captureCallback(void *inRefCon,
AudioUnitRenderActionFlags *ioActionFlags,
const AudioTimeStamp *inTimeStamp,
UInt32 inBusNumber,
UInt32 inNumberFrames,
AudioBufferList *ioData)
{
MacOS_AudioDevice* _this = static_cast<MacOS_AudioDevice*>(inRefCon);
// Get the new audio data
OSStatus err = AudioUnitRender(_this->m_AUHAL, ioActionFlags, inTimeStamp, inBusNumber, inNumberFrames, _this->m_InputBuffer);
if (err != noErr)
{
...
return err;
}
// ignore callback on unexpected buffer size
if (_this->m_params.bufferSizeSamples != inNumberFrames)
{
...
return noErr;
}
// Deliver audio data
DeviceIOMessage message;
message.bufferSizeBytes = _this->m_deviceBufferSizeBytes;
message.buffer = _this->m_InputBuffer->mBuffers[0].mData;
if (_this->m_callbackFunc)
{
_this->m_callbackFunc(_this, message);
}
}
打开，并开始捕捉设备：
void MacOS_AudioDevice::openAUHALCapture()
{
UInt32 enableIO;
AudioStreamBasicDescription streamFormat;
UInt32 size;
SInt32 *channelArr;
std::stringstream ss;
AudioObjectPropertyAddress deviceBufSizeProperty =
{
kAudioDevicePropertyBufferFrameSize,
kAudioDevicePropertyScopeInput,
kAudioObjectPropertyElementMaster
};
// AUHAL
AudioComponentDescription cd = {kAudioUnitType_Output, kAudioUnitSubType_HALOutput, kAudioUnitManufacturer_Apple, 0, 0};
AudioComponent HALOutput = AudioComponentFindNext(NULL, &cd);
verify_macosapi(AudioComponentInstanceNew(HALOutput, &m_AUHAL));
verify_macosapi(AudioUnitInitialize(m_AUHAL));
// enable input IO
enableIO = 1;
verify_macosapi(AudioUnitSetProperty(m_AUHAL, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(enableIO)));
// disable output IO
enableIO = 0;
verify_macosapi(AudioUnitSetProperty(m_AUHAL, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(enableIO)));
// Setup current device
size = sizeof(AudioDeviceID);
verify_macosapi(AudioUnitSetProperty(m_AUHAL, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &m_MacDeviceID, sizeof(AudioDeviceID)));
// Set device native buffer length before setting AUHAL stream
size = sizeof(m_originalDeviceBufferTimeFrames);
verify_macosapi(AudioObjectSetPropertyData(m_MacDeviceID, &deviceBufSizeProperty, 0, NULL, size, &m_originalDeviceBufferTimeFrames));
// Get device format
size = sizeof(AudioStreamBasicDescription);
verify_macosapi(AudioUnitGetProperty(m_AUHAL, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &streamFormat, &size));
// Setup channel map
assert(m_params.numOfChannels <= streamFormat.mChannelsPerFrame);
channelArr = new SInt32[streamFormat.mChannelsPerFrame];
for (int i = 0; i < streamFormat.mChannelsPerFrame; i++)
channelArr[i] = -1;
for (int i = 0; i < m_params.numOfChannels; i++)
channelArr[i] = i;
verify_macosapi(AudioUnitSetProperty(m_AUHAL, kAudioOutputUnitProperty_ChannelMap, kAudioUnitScope_Input, 1, channelArr, sizeof(SInt32) * streamFormat.mChannelsPerFrame));
delete [] channelArr;
// Setup stream converters
streamFormat.mFormatID = kAudioFormatLinearPCM;
streamFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger;
streamFormat.mFramesPerPacket = m_SamplesPerPacket;
streamFormat.mBitsPerChannel = m_params.sampleDepthBits;
streamFormat.mSampleRate = m_deviceSampleRate;
streamFormat.mChannelsPerFrame = 1;
streamFormat.mBytesPerFrame = 2;
streamFormat.mBytesPerPacket = streamFormat.mFramesPerPacket * streamFormat.mBytesPerFrame;
verify_macosapi(AudioUnitSetProperty(m_AUHAL, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, &streamFormat, size));
// Setup callbacks
AURenderCallbackStruct input;
input.inputProc = captureCallback;
input.inputProcRefCon = this;
verify_macosapi(AudioUnitSetProperty(m_AUHAL, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(input)));
// Calculate the size of the IO buffer (in samples)
if (m_params.bufferSizeMS != -1)
{
unsigned int desiredSignalsInBuffer = (m_params.bufferSizeMS/(double)1000) * m_deviceSampleRate;
// making sure the value stay in the device's supported range
desiredSignalsInBuffer = std::min<unsigned int>(desiredSignalsInBuffer, m_deviceBufferFramesRange.mMaximum);
desiredSignalsInBuffer = std::max<unsigned int>(m_deviceBufferFramesRange.mMinimum, desiredSignalsInBuffer);
m_deviceBufferFrames = desiredSignalsInBuffer;
}
// Set device buffer length
size = sizeof(m_deviceBufferFrames);
verify_macosapi(AudioObjectSetPropertyData(m_MacDeviceID, &deviceBufSizeProperty, 0, NULL, size, &m_deviceBufferFrames));
m_deviceBufferSizeBytes = m_deviceBufferFrames * streamFormat.mBytesPerFrame;
m_deviceBufferTimeMS = 1000 * m_deviceBufferFrames/m_deviceSampleRate;
// Calculate number of buffers from channels
size = offsetof(AudioBufferList, mBuffers[0]) + (sizeof(AudioBuffer) * m_params.numOfChannels);
// Allocate input buffer
m_InputBuffer = (AudioBufferList *)malloc(size);
m_InputBuffer->mNumberBuffers = m_params.numOfChannels;
// Pre-malloc buffers for AudioBufferLists
for(UInt32 i = 0; i< m_InputBuffer->mNumberBuffers ; i++)
{
m_InputBuffer->mBuffers[i].mNumberChannels = 1;
m_InputBuffer->mBuffers[i].mDataByteSize = m_deviceBufferSizeBytes;
m_InputBuffer->mBuffers[i].mData = malloc(m_deviceBufferSizeBytes);
}
// Update class properties
m_params.sampleRateHz = streamFormat.mSampleRate;
m_params.bufferSizeSamples = m_deviceBufferFrames;
m_params.bufferSizeBytes = m_params.bufferSizeSamples * streamFormat.mBytesPerFrame;
}
eADMReturnCode MacOS_AudioDevice::start()
{
eADMReturnCode ret = OK;
LOGAPI(ret);
if (!m_isStarted && m_isOpen)
{
OSStatus err = AudioOutputUnitStart(m_AUHAL);
if (err == noErr)
m_isStarted = true;
else
ret = ERROR;
}
return ret;
}
任何想法是什么原因，以及如何解决？
在此先感谢！
请显示您正在使用的过程。 – dave234
我用代码更新了我的问题。请参阅上文。谢谢！ – meirm
看起来hotpaw2对我来说很合适。 – dave234