Add module: speaker capture

test/audio_capture/audio_capture_wasapi.cpp

@@ -0,0 +1,303 @@
+// MyAudioSink.cpp : 定义控制台应用程序的入口点。
+//
+
+// #define  _CRT_SECURE_NO_WARNINGS
+
+#include <Audioclient.h>
+#include <Devicetopology.h>
+#include <Endpointvolume.h>
+#include <Mmdeviceapi.h>
+#include <tchar.h>
+
+#include <iostream>
+//-----------------------------------------------------------
+// Record an audio stream from the default audio capture
+// device. The RecordAudioStream function allocates a shared
+// buffer big enough to hold one second of PCM audio data.
+// The function uses this buffer to stream data from the
+// capture device. The main loop runs every 1/2 second.
+//-----------------------------------------------------------
+
+// REFERENCE_TIME time units per second and per millisecond
+#define REFTIMES_PER_SEC 10000000
+#define REFTIMES_PER_MILLISEC 10000
+
+#define EXIT_ON_ERROR(hres) \
+  if (FAILED(hres)) {       \
+    goto Exit;              \
+  }
+
+#define SAFE_RELEASE(punk) \
+  if ((punk) != NULL) {    \
+    (punk)->Release();     \
+    (punk) = NULL;         \
+  }
+
+#define IS_INPUT_DEVICE 0  // 切换输入和输出音频设备
+
+#define BUFFER_TIME_100NS (5 * 10000000)
+
+const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
+const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
+const IID IID_IAudioClient = __uuidof(IAudioClient);
+const IID IID_IAudioCaptureClient = __uuidof(IAudioCaptureClient);
+
+const IID IID_IDeviceTopology = __uuidof(IDeviceTopology);
+const IID IID_IAudioVolumeLevel = __uuidof(IAudioVolumeLevel);
+const IID IID_IPart = __uuidof(IPart);
+const IID IID_IConnector = __uuidof(IConnector);
+const IID IID_IAudioEndpointVolume = __uuidof(IAudioEndpointVolume);
+
+class MyAudioSink {
+ public:
+  // WAVEFORMATEX *pwfx = NULL;
+  int SetFormat(WAVEFORMATEX *pwfx);
+
+  int CopyData(SHORT *pData, UINT32 numFramesAvailable, BOOL *pbDone);
+};
+
+int MyAudioSink::SetFormat(WAVEFORMATEX *pwfx) {
+  printf("wFormatTag is %x\n", pwfx->wFormatTag);
+  printf("nChannels is %x\n", pwfx->nChannels);
+  printf("nSamplesPerSec is %d\n", pwfx->nSamplesPerSec);
+  printf("nAvgBytesPerSec is %d\n", pwfx->nAvgBytesPerSec);
+  printf("wBitsPerSample is %d\n", pwfx->wBitsPerSample);
+
+  return 0;
+}
+
+FILE *fp;
+
+int MyAudioSink::CopyData(SHORT *pData, UINT32 numFramesAvailable,
+                          BOOL *pbDone) {
+  if (pData != NULL) {
+    size_t t = sizeof(SHORT);
+    for (int i = 0; i < numFramesAvailable / t; i++) {
+      double dbVal = pData[i];
+      pData[i] = dbVal;  // 可以通过不同的分母来控制声音大小
+    }
+    fwrite(pData, numFramesAvailable, 1, fp);
+  }
+
+  return 0;
+}
+
+/// pwfx->nSamplesPerSec    = 44100;
+/// 不支持修改采样率， 看来只能等得到数据之后再 swr 转换了
+BOOL AdjustFormatTo16Bits(WAVEFORMATEX *pwfx) {
+  BOOL bRet(FALSE);
+
+  if (pwfx->wFormatTag == WAVE_FORMAT_IEEE_FLOAT) {
+    pwfx->wFormatTag = WAVE_FORMAT_PCM;
+    pwfx->wBitsPerSample = 16;
+    pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
+    pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
+
+    bRet = TRUE;
+  } else if (pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
+    PWAVEFORMATEXTENSIBLE pEx = reinterpret_cast<PWAVEFORMATEXTENSIBLE>(pwfx);
+    if (IsEqualGUID(KSDATAFORMAT_SUBTYPE_IEEE_FLOAT, pEx->SubFormat)) {
+      pEx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
+      pEx->Samples.wValidBitsPerSample = 16;
+      pwfx->wBitsPerSample = 16;
+      pwfx->nBlockAlign = pwfx->nChannels * pwfx->wBitsPerSample / 8;
+      pwfx->nAvgBytesPerSec = pwfx->nBlockAlign * pwfx->nSamplesPerSec;
+
+      bRet = TRUE;
+    }
+  }
+
+  return bRet;
+}
+
+typedef unsigned long long uint64_t;
+static bool have_clockfreq = false;
+static LARGE_INTEGER clock_freq;
+static inline uint64_t get_clockfreq(void) {
+  if (!have_clockfreq) QueryPerformanceFrequency(&clock_freq);
+  return clock_freq.QuadPart;
+}
+uint64_t os_gettime_ns(void) {
+  LARGE_INTEGER current_time;
+  double time_val;
+
+  QueryPerformanceCounter(&current_time);
+  time_val = (double)current_time.QuadPart;
+  time_val *= 1000000000.0;
+  time_val /= (double)get_clockfreq();
+
+  return (uint64_t)time_val;
+}
+
+HRESULT RecordAudioStream(MyAudioSink *pMySink) {
+  HRESULT hr;
+  REFERENCE_TIME hnsActualDuration;
+  UINT32 bufferFrameCount;
+  UINT32 numFramesAvailable;
+  BYTE *pData;
+  DWORD flags;
+  REFERENCE_TIME hnsDefaultDevicePeriod(0);
+
+  REFERENCE_TIME hnsRequestedDuration = REFTIMES_PER_SEC;
+  IMMDeviceEnumerator *pEnumerator = NULL;
+  IMMDevice *pDevice = NULL;
+  IAudioClient *pAudioClient = NULL;
+  IAudioCaptureClient *pCaptureClient = NULL;
+  WAVEFORMATEX *pwfx = NULL;
+  UINT32 packetLength = 0;
+  BOOL bDone = FALSE;
+  HANDLE hTimerWakeUp = NULL;
+  UINT64 pos, ts;
+
+  hr = CoCreateInstance(CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
+                        IID_IMMDeviceEnumerator, (void **)&pEnumerator);
+
+  EXIT_ON_ERROR(hr)
+
+  if (IS_INPUT_DEVICE)
+    hr = pEnumerator->GetDefaultAudioEndpoint(eCapture, eCommunications,
+                                              &pDevice);  // 输入
+  else
+    hr = pEnumerator->GetDefaultAudioEndpoint(eRender, eConsole,
+                                              &pDevice);  // 输出
+
+  // wchar_t *w_id;
+  // os_utf8_to_wcs_ptr(device_id.c_str(), device_id.size(), &w_id);
+  // hr = pEnumerator->GetDevice(w_id, &pDevice);
+  // bfree(w_id);
+
+  EXIT_ON_ERROR(hr)
+
+  hr = pDevice->Activate(IID_IAudioClient, CLSCTX_ALL, NULL,
+                         (void **)&pAudioClient);
+
+  EXIT_ON_ERROR(hr)
+
+  hr = pAudioClient->GetMixFormat(&pwfx);
+
+  EXIT_ON_ERROR(hr)
+
+  // The GetDevicePeriod method retrieves the length of the periodic interval
+  // separating successive processing passes by the audio engine on the data in
+  // the endpoint buffer.
+  hr = pAudioClient->GetDevicePeriod(&hnsDefaultDevicePeriod, NULL);
+
+  EXIT_ON_ERROR(hr)
+
+  AdjustFormatTo16Bits(pwfx);
+
+  // 平时创建定时器使用的是WINAPI  SetTimer，不过该函数一般用于有界面的时候。
+  // 无界面的情况下，可以选择微软提供的CreateWaitableTimer和SetWaitableTimer
+  // API。
+  hTimerWakeUp = CreateWaitableTimer(NULL, FALSE, NULL);
+
+  DWORD flag;
+  if (IS_INPUT_DEVICE)
+    flag = 0;
+  else
+    flag = AUDCLNT_STREAMFLAGS_LOOPBACK;
+
+  if (IS_INPUT_DEVICE)
+    hr = pAudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, flag /*0*/, 0, 0,
+                                  pwfx, NULL);  // 输入
+  else
+    hr = pAudioClient->Initialize(AUDCLNT_SHAREMODE_SHARED, flag /*0*/, 0, 0,
+                                  pwfx, NULL);  // 输出
+
+  EXIT_ON_ERROR(hr)
+
+  // Get the size of the allocated buffer.
+  hr = pAudioClient->GetBufferSize(&bufferFrameCount);
+  EXIT_ON_ERROR(hr)
+
+  hr = pAudioClient->GetService(IID_IAudioCaptureClient,
+                                (void **)&pCaptureClient);
+
+  EXIT_ON_ERROR(hr)
+
+  LARGE_INTEGER liFirstFire;
+  liFirstFire.QuadPart =
+      -hnsDefaultDevicePeriod / 2;  // negative means relative time
+  LONG lTimeBetweenFires = (LONG)hnsDefaultDevicePeriod / 2 /
+                           (10 * 1000);  // convert to milliseconds
+
+  BOOL bOK = SetWaitableTimer(hTimerWakeUp, &liFirstFire, lTimeBetweenFires,
+                              NULL, NULL, FALSE);
+
+  // Notify the audio sink which format to use.
+  hr = pMySink->SetFormat(pwfx);
+  EXIT_ON_ERROR(hr)
+
+  // Calculate the actual duration of the allocated buffer.
+  hnsActualDuration =
+      (double)REFTIMES_PER_SEC * bufferFrameCount / pwfx->nSamplesPerSec;
+
+  /*************************************************************/
+  hr = pAudioClient->Start();  // Start recording.
+  EXIT_ON_ERROR(hr)
+  HANDLE waitArray[1] = {/*htemp hEventStop,*/ hTimerWakeUp};
+
+  // Each loop fills about half of the shared buffer.
+  while (bDone == FALSE) {
+    // Sleep for half the buffer duration.
+    // Sleep(hnsActualDuration/REFTIMES_PER_MILLISEC/2);//这句貌似不加也可以
+    // WaitForSingleObject(hTimerWakeUp,INFINITE);
+    int a = sizeof(waitArray);
+    int aa = sizeof(waitArray[0]);
+    WaitForMultipleObjects(sizeof(waitArray) / sizeof(waitArray[0]), waitArray,
+                           FALSE, INFINITE);
+    // WaitForMultipleObjects(sizeof(waitArray) / sizeof(waitArray[0]),
+    // waitArray, FALSE, INFINITE);
+
+    hr = pCaptureClient->GetNextPacketSize(&packetLength);
+    EXIT_ON_ERROR(hr)
+
+    while (packetLength != 0) {
+      // Get the available data in the shared buffer.
+      hr = pCaptureClient->GetBuffer(&pData, &numFramesAvailable, &flags, NULL,
+                                     &ts);
+      ts = ts * 100;
+      uint64_t timestamp =
+          os_gettime_ns();  // ts是设备时间，timestamp是系统时间
+      EXIT_ON_ERROR(hr)
+
+      // 位运算，flags的标志符为2（静音状态）时，将pData置为NULL
+      if (flags & AUDCLNT_BUFFERFLAGS_SILENT) {
+        pData = NULL;  // Tell CopyData to write silence.
+      }
+
+      // Copy the available capture data to the audio sink.
+      hr = pMySink->CopyData((SHORT *)pData,
+                             numFramesAvailable * pwfx->nBlockAlign, &bDone);
+      EXIT_ON_ERROR(hr)
+
+      hr = pCaptureClient->ReleaseBuffer(numFramesAvailable);
+      EXIT_ON_ERROR(hr)
+
+      hr = pCaptureClient->GetNextPacketSize(&packetLength);
+      EXIT_ON_ERROR(hr)
+    }
+  }
+
+  hr = pAudioClient->Stop();  // Stop recording.
+  EXIT_ON_ERROR(hr)
+
+Exit:
+  CoTaskMemFree(pwfx);
+  SAFE_RELEASE(pEnumerator)
+  SAFE_RELEASE(pDevice)
+  SAFE_RELEASE(pAudioClient)
+  SAFE_RELEASE(pCaptureClient)
+
+  return hr;
+}
+
+int _tmain(int argc, _TCHAR *argv[]) {
+  fopen_s(&fp, "record.pcm", "wb");
+  CoInitialize(NULL);
+  MyAudioSink test;
+
+  RecordAudioStream(&test);
+
+  return 0;
+}