提供一點上下文。我正嘗試從c#應用程序中的相機輸出實時音頻。做了一些研究後,在C++託管的dll中執行它似乎很明顯。我選擇了XAudio2 API,因爲它應該很容易實現和使用動態音頻內容。XAudio2 - 使用動態緩衝區時打開輸出
所以這個想法是用C++創建一個空緩衝區的XAudio設備,並從C#代碼端推入音頻。音頻塊每隔50ms被推送一次,因爲我希望儘可能縮短延遲。定時器運行它
// SampleRate = 44100; Channels = 2; BitPerSample = 16;
var blockAlign = (Channels * BitsPerSample)/8;
var avgBytesPerSecond = SampleRate * blockAlign;
var avgBytesPerMillisecond = avgBytesPerSecond/1000;
var bufferSize = avgBytesPerMillisecond * Time;
_sampleBuffer = new byte[bufferSize];
每次獲取音頻緩衝器的指針,從音頻讀取數據,將數據複製到指針並調用PushAudio方法。 我也使用秒錶來檢查處理過程花了多長時間,並再次計算定時器的時間間隔以包含處理時間。
private void PushAudioChunk(object sender, ElapsedEventArgs e)
{
unsafe
{
_pushAudioStopWatch.Reset();
_pushAudioStopWatch.Start();
var audioBufferPtr = Output.AudioCapturerBuffer();
FillBuffer(_sampleBuffer);
Marshal.Copy(_sampleBuffer, 0, (IntPtr)audioBufferPtr, _sampleBuffer.Length);
Output.PushAudio();
_pushTimer.Interval = Time - _pushAudioStopWatch.ElapsedMilliseconds;
_pushAudioStopWatch.Stop();
DIX.Log.WriteLine("Push audio took: {0}ms", _pushAudioStopWatch.ElapsedMilliseconds);
}
}
這是C++部分的實現。
關於msdn的文檔,我創建了一個XAudio2設備並添加了MasterVoice和SourceVoice。起初緩衝區是空的,因爲c#部分負責推入音頻數據。
namespace Audio
{
using namespace System;
template <class T> void SafeRelease(T **ppT)
{
if (*ppT)
{
(*ppT)->Release();
*ppT = NULL;
}
}
WAVEFORMATEXTENSIBLE wFormat;
XAUDIO2_BUFFER buffer = { 0 };
IXAudio2* pXAudio2 = NULL;
IXAudio2MasteringVoice* pMasterVoice = NULL;
IXAudio2SourceVoice* pSourceVoice = NULL;
WaveOut::WaveOut(int bufferSize)
{
audioBuffer = new Byte[bufferSize];
wFormat.Format.wFormatTag = WAVE_FORMAT_PCM;
wFormat.Format.nChannels = 2;
wFormat.Format.nSamplesPerSec = 44100;
wFormat.Format.wBitsPerSample = 16;
wFormat.Format.nBlockAlign = (wFormat.Format.nChannels * wFormat.Format.wBitsPerSample)/8;
wFormat.Format.nAvgBytesPerSec = wFormat.Format.nSamplesPerSec * wFormat.Format.nBlockAlign;
wFormat.Format.cbSize = 0;
wFormat.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
HRESULT hr = XAudio2Create(&pXAudio2, 0, XAUDIO2_DEFAULT_PROCESSOR);
if (SUCCEEDED(hr))
{
hr = pXAudio2->CreateMasteringVoice(&pMasterVoice);
}
if (SUCCEEDED(hr))
{
hr = pXAudio2->CreateSourceVoice(&pSourceVoice, (WAVEFORMATEX*)&wFormat,
0, XAUDIO2_DEFAULT_FREQ_RATIO, NULL, NULL, NULL);
}
buffer.pAudioData = (BYTE*)audioBuffer;
buffer.AudioBytes = bufferSize;
buffer.Flags = 0;
if (SUCCEEDED(hr))
{
hr = pSourceVoice->Start(0);
}
}
WaveOut::~WaveOut()
{
}
WaveOut^ WaveOut::CreateWaveOut(int bufferSize)
{
return gcnew WaveOut(bufferSize);
}
uint8_t* WaveOut::AudioCapturerBuffer()
{
if (!audioBuffer)
{
throw gcnew Exception("Audio buffer is not initialized. Did you forget to set up the audio container?");
}
return (BYTE*)audioBuffer;
}
int WaveOut::PushAudio()
{
HRESULT hr = pSourceVoice->SubmitSourceBuffer(&buffer);
if (FAILED(hr))
{
return -1;
}
return 0;
}
}
我面臨的問題是,我總是有一些輸出開裂。我試圖增加定時器的間隔或增加緩衝區大小。每次都有相同的結果。
我在做什麼錯?
更新:
我在3個緩衝區XAudio引擎可以順利通過。裂開了。現在缺少的部分是在c#部分的正確時間填充緩衝區,以避免使用相同數據的緩衝區。
void Render(void* param)
{
std::vector<byte> audioBuffers[BUFFER_COUNT];
size_t currentBuffer = 0;
// Get the current state of the source voice
while (BackgroundThreadRunning && pSourceVoice)
{
if (pSourceVoice)
{
pSourceVoice->GetState(&state);
}
while (state.BuffersQueued < BUFFER_COUNT)
{
std::vector<byte> resultData;
resultData.resize(DATA_SIZE);
CopyMemory(&resultData[0], pAudioBuffer, DATA_SIZE);
// Retreive the next buffer to stream from MF Music Streamer
audioBuffers[currentBuffer] = resultData;
// Submit the new buffer
XAUDIO2_BUFFER buf = { 0 };
buf.AudioBytes = static_cast<UINT32>(audioBuffers[currentBuffer].size());
buf.pAudioData = &audioBuffers[currentBuffer][0];
pSourceVoice->SubmitSourceBuffer(&buf);
// Advance the buffer index
currentBuffer = ++currentBuffer % BUFFER_COUNT;
// Get the updated state
pSourceVoice->GetState(&state);
}
Sleep(30);
}
}
在黑暗中野生刺傷 - 你有沒有嘗試清空緩衝區? – BugFinder
你好。不,我沒有。我應該怎麼做?在推新音頻塊之前? – datoml
定義它之後,只需將每個字節設置爲0 – BugFinder