void nuiAudioDevice_DS_ProcessingTh::Process(uint pos)
{
NGL_LOG(_T("nuiAudioDevice_DS_ProcessingTh"), NGL_LOG_DEBUG, _T("Process Thread received Event %d\n"), pos);
int16* pBuf1=NULL;
int16* pBuf2=NULL;
DWORD size1=0;
DWORD size2=0;
DWORD bufferBytes = mBufferSize * mInputNbChannels * sizeof(int16);
if (!mpRingBuffer->GetWritable())
return;
//
// lock the input buffer if any,
// and read data from it to the local buffer
//
if ( mpDSInputBuffer
&& mpDSInputBuffer->Lock(pos * bufferBytes /* offset */, bufferBytes /*size*/, (LPVOID*)&pBuf1, &size1, (LPVOID*)&pBuf2, &size2, 0)
)
{
if (!pBuf1 || !size1)
{
//NGL_LOG(_T("nuiAudioDevice_DS_ProcessingTh"), NGL_LOG_ERROR, _T("Process error : could not lock any part of the input buffer\n"));
NGL_ASSERT(0);
return;
}
// check that we got the right size
NGL_ASSERT((size1+size2) == bufferBytes);
// copy input data into local buffer
memcpy(mpLocalBuf, pBuf1, size1);
if (pBuf2)
memcpy(mpLocalBuf+size1, pBuf2, size2);
// convert int16 buffer to float buffer
for (uint32 ch=0; ch < mInputNbChannels; ch++)
{
nuiAudioConvert_INint16ToDEfloat(mpLocalBuf, const_cast<float*>(mFloatInputBuf[ch]), ch, mInputNbChannels, mBufferSize);
}
}
// call user audio process function
mpProcessFunction(mFloatInputBuf, mFloatOutputBuf, mBufferSize);
// copy output buffer contents to ringbuffer
uint32 nbWrite = WriteToRingBuf(mFloatOutputBuf, mBufferSize, mOutputNbChannels);
// release DS input buffer
if (mpDSInputBuffer)
mpDSInputBuffer->Unlock(pBuf1, size1, pBuf2, size2);
}
void DXAudioInput::run() {
LPDIRECTSOUNDCAPTURE8 pDSCapture;
LPDIRECTSOUNDCAPTUREBUFFER pDSCaptureBuffer;
LPDIRECTSOUNDNOTIFY8 pDSNotify;
DWORD dwBufferSize;
bool bOk;
DWORD dwReadyBytes = 0;
DWORD dwLastReadPos = 0;
DWORD dwReadPosition;
DWORD dwCapturePosition;
LPVOID aptr1, aptr2;
DWORD nbytes1, nbytes2;
HRESULT hr;
WAVEFORMATEX wfx;
DSCBUFFERDESC dscbd;
pDSCapture = NULL;
pDSCaptureBuffer = NULL;
pDSNotify = NULL;
bOk = false;
bool failed = false;
float safety = 2.0f;
bool didsleep = false;
bool firstsleep = false;
Timer t;
ZeroMemory(&wfx, sizeof(wfx));
wfx.wFormatTag = WAVE_FORMAT_PCM;
ZeroMemory(&dscbd, sizeof(dscbd));
dscbd.dwSize = sizeof(dscbd);
dscbd.dwBufferBytes = dwBufferSize = iFrameSize * sizeof(short) * NBUFFBLOCKS;
dscbd.lpwfxFormat = &wfx;
wfx.nChannels = 1;
wfx.nSamplesPerSec = iSampleRate;
wfx.nBlockAlign = 2;
wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
wfx.wBitsPerSample = 16;
// Create IDirectSoundCapture using the preferred capture device
if (! g.s.qbaDXInput.isEmpty()) {
LPGUID lpguid = reinterpret_cast<LPGUID>(g.s.qbaDXInput.data());
if (FAILED(hr = DirectSoundCaptureCreate8(lpguid, &pDSCapture, NULL))) {
failed = true;
}
}
if (! pDSCapture && FAILED(hr = DirectSoundCaptureCreate8(&DSDEVID_DefaultVoiceCapture, &pDSCapture, NULL)))
qWarning("DXAudioInput: DirectSoundCaptureCreate failed: hr=0x%08lx", hr);
else if (FAILED(hr = pDSCapture->CreateCaptureBuffer(&dscbd, &pDSCaptureBuffer, NULL)))
qWarning("DXAudioInput: CreateCaptureBuffer failed: hr=0x%08lx", hr);
else if (FAILED(hr = pDSCaptureBuffer->QueryInterface(IID_IDirectSoundNotify, reinterpret_cast<void **>(&pDSNotify))))
qWarning("DXAudioInput: QueryInterface (Notify) failed: hr=0x%08lx", hr);
else
bOk = true;
if (failed)
g.mw->msgBox(tr("Opening chosen DirectSound Input failed. Default device will be used."));
qWarning("DXAudioInput: Initialized");
if (! bOk)
goto cleanup;
if (FAILED(hr = pDSCaptureBuffer->Start(DSCBSTART_LOOPING))) {
qWarning("DXAudioInput: Start failed: hr=0x%08lx", hr);
} else {
while (bRunning) {
firstsleep = true;
didsleep = false;
do {
if (FAILED(hr = pDSCaptureBuffer->GetCurrentPosition(&dwCapturePosition, &dwReadPosition))) {
qWarning("DXAudioInput: GetCurrentPosition failed: hr=0x%08lx", hr);
bRunning = false;
break;
}
if (dwReadPosition < dwLastReadPos)
dwReadyBytes = (dwBufferSize - dwLastReadPos) + dwReadPosition;
else
dwReadyBytes = dwReadPosition - dwLastReadPos;
if (static_cast<int>(dwReadyBytes) < sizeof(short) * iFrameSize) {
double msecleft = 20.0 - (dwReadyBytes * 20.0) / (sizeof(short) * iFrameSize);
if (didsleep)
safety *= 1.1f;
else if (firstsleep)
safety *= 0.998f;
int msec = static_cast<int>(msecleft + (firstsleep ? safety : 0.0));
msleep(msec);
didsleep = true;
firstsleep = false;
}
} while (static_cast<int>(dwReadyBytes) < sizeof(short) * iFrameSize);
// Desynchonized?
if (dwReadyBytes > (dwBufferSize / 2)) {
qWarning("DXAudioInput: Lost synchronization");
dwLastReadPos = dwReadPosition;
} else if (bRunning) {
if (FAILED(hr = pDSCaptureBuffer->Lock(dwLastReadPos, sizeof(short) * iFrameSize, &aptr1, &nbytes1, &aptr2, &nbytes2, 0))) {
qWarning("DXAudioInput: Lock from %ld (%d bytes) failed: hr=0x%08lx",dwLastReadPos, sizeof(short) * iFrameSize, hr);
bRunning = false;
break;
}
if (aptr1 && nbytes1)
CopyMemory(psMic, aptr1, nbytes1);
if (aptr2 && nbytes2)
CopyMemory(psMic+nbytes1/2, aptr2, nbytes2);
if (FAILED(hr = pDSCaptureBuffer->Unlock(aptr1, nbytes1, aptr2, nbytes2))) {
qWarning("DXAudioInput: Unlock failed: hr=0x%08lx", hr);
bRunning = false;
break;
}
dwLastReadPos = (dwLastReadPos + sizeof(short) * iFrameSize) % dwBufferSize;
encodeAudioFrame();
}
}
if (! FAILED(hr))
pDSCaptureBuffer->Stop();
}
if (FAILED(hr)) {
g.mw->msgBox(tr("Lost DirectSound input device."));
}
cleanup:
if (! bOk) {
g.mw->msgBox(tr("Opening chosen DirectSound Input device failed. No microphone capture will be done."));
}
if (pDSNotify)
pDSNotify->Release();
if (pDSCaptureBuffer)
pDSCaptureBuffer->Release();
if (pDSCapture)
pDSCapture->Release();
}
uae_u8 sampler_getsample (int channel)
{
#if 0
int cur_pos;
static int cap_pos;
static float diffsample;
#endif
static double doffset_offset;
HRESULT hr;
DWORD t;
void *p1, *p2;
DWORD len1, len2;
evt cycles;
int sample, samplecnt;
double doffset;
int offset;
if (!currprefs.sampler_stereo)
channel = 0;
if (!inited) {
DWORD pos;
if (!capture_init ()) {
capture_free ();
return 0;
}
inited = 1;
oldcycles = get_cycles ();
oldoffset = -1;
doffset_offset = 0;
hr = lpDSB2r->GetCurrentPosition (&t, &pos);
if (FAILED (hr)) {
sampler_free ();
return 0;
}
if (t >= pos)
safediff = t - pos;
else
safediff = recordbufferframes * SAMPLESIZE - pos + t;
write_log (_T("SAMPLER: safediff %d %d\n"), safediff, safediff + sampleframes * SAMPLESIZE);
safediff += 4 * sampleframes * SAMPLESIZE;
#if 0
diffsample = 0;
safepos = -recordbufferframes / 10 * SAMPLESIZE;
hr = lpDSB2r->GetCurrentPosition (&t, &pos);
cap_pos = pos;
cap_pos += safepos;
if (cap_pos < 0)
cap_pos += recordbufferframes * SAMPLESIZE;
if (cap_pos >= recordbufferframes * SAMPLESIZE)
cap_pos -= recordbufferframes * SAMPLESIZE;
if (FAILED (hr)) {
sampler_free ();
return 0;
}
#endif
}
if (clockspersample < 1)
return 0;
uae_s16 *sbuf = (uae_s16*)samplebuffer;
vsynccnt = 0;
sample = 0;
samplecnt = 0;
cycles = (int)get_cycles () - (int)oldcycles;
doffset = doffset_offset + cycles / clockspersample;
offset = (int)doffset;
if (oldoffset < 0 || offset >= sampleframes || offset < 0) {
if (offset >= sampleframes) {
doffset -= offset;
doffset_offset = doffset;
}
if (oldoffset >= 0 && offset >= sampleframes) {
while (oldoffset < sampleframes) {
sample += sbuf[oldoffset * SAMPLESIZE / 2 + channel];
oldoffset++;
samplecnt++;
}
}
hr = lpDSB2r->GetCurrentPosition (&t, NULL);
int pos = t;
pos -= safediff;
if (pos < 0)
pos += recordbufferframes * SAMPLESIZE;
hr = lpDSB2r->Lock (pos, sampleframes * SAMPLESIZE, &p1, &len1, &p2, &len2, 0);
if (FAILED (hr)) {
write_log (_T("SAMPLER: Lock() failed %x\n"), hr);
return 0;
}
memcpy (samplebuffer, p1, len1);
if (p2)
memcpy (samplebuffer + len1, p2, len2);
lpDSB2r->Unlock (p1, len1, p2, len2);
#if 0
cap_pos = t;
cap_pos += sampleframes * SAMPLESIZE;
if (cap_pos < 0)
cap_pos += RECORDBUFFER * SAMPLESIZE;
if (cap_pos >= RECORDBUFFER * SAMPLESIZE)
cap_pos -= RECORDBUFFER * SAMPLESIZE;
hr = lpDSB2r->GetCurrentPosition (&t, &pos);
cur_pos = pos;
if (FAILED (hr))
return 0;
cur_pos += safepos;
if (cur_pos < 0)
cur_pos += RECORDBUFFER * SAMPLESIZE;
if (cur_pos >= RECORDBUFFER * SAMPLESIZE)
cur_pos -= RECORDBUFFER * SAMPLESIZE;
int diff;
if (cur_pos >= cap_pos)
diff = cur_pos - cap_pos;
else
diff = RECORDBUFFER * SAMPLESIZE - cap_pos + cur_pos;
if (diff > RECORDBUFFER * SAMPLESIZE / 2)
diff -= RECORDBUFFER * SAMPLESIZE;
diff /= SAMPLESIZE;
int diff2 = 100 * diff / (RECORDBUFFER / 2);
diffsample = -pow (diff2 < 0 ? -diff2 : diff2, 3.1);
if (diff2 < 0)
diffsample = -diffsample;
write_log (_T("%d\n"), diff);
write_log (_T("CAP=%05d CUR=%05d (%-05d) OFF=%05d %f\n"),
cap_pos / SAMPLESIZE, cur_pos / SAMPLESIZE, (cap_pos - cur_pos) / SAMPLESIZE, offset, doffset_offset);
#endif
if (offset < 0)
offset = 0;
if (offset >= sampleframes)
offset -= sampleframes;
oldoffset = 0;
oldcycles = get_cycles ();
}
while (oldoffset <= offset) {
sample += sbuf[oldoffset * SAMPLESIZE / 2 + channel];
samplecnt++;
oldoffset++;
}
oldoffset = offset;
if (samplecnt > 0)
sample /= samplecnt;
#if 1
/* yes, not 256, without this max recording volume would still be too quiet on my sound cards */
sample /= 128;
if (sample < -128)
sample = 0;
else if (sample > 127)
sample = 127;
return (uae_u8)(sample - 128);
#else
return (Uae_u8)((sample / 256) - 128);
#endif
}
int VoiceRecord_DSound::GetRecordedData(short *pOut, int nSamplesInt)
{
if(!m_pCaptureBuffer)
{
assert(false);
return 0;
}
DWORD dwStatus;
HRESULT hr = m_pCaptureBuffer->GetStatus(&dwStatus);
if(FAILED(hr) || !(dwStatus & DSCBSTATUS_CAPTURING))
return 0;
Idle(); // Update wrapping..
DWORD nSamplesWanted = (DWORD)nSamplesInt;
DWORD dwCapturePos, dwReadPos;
hr = m_pCaptureBuffer->GetCurrentPosition(&dwCapturePos, &dwReadPos);
if(FAILED(hr))
return 0;
dwCapturePos += m_WrapOffset;
dwReadPos += m_WrapOffset;
// Read the range (dwReadPos-nSamplesWanted, dwReadPos), but don't re-read data we've already read.
DWORD readStart;
if(dwReadPos > nSamplesWanted)
readStart = dwReadPos - nSamplesWanted;
else
readStart = 0;
if(readStart < m_LastReadPos)
readStart = m_LastReadPos;
// Lock the buffer.
LPVOID pData[2];
DWORD dataLen[2];
hr = m_pCaptureBuffer->Lock(
readStart % NumCaptureBufferBytes(), // Offset.
dwReadPos - readStart, // Number of bytes to lock.
&pData[0], // Buffer 1.
&dataLen[0], // Buffer 1 length.
&pData[1], // Buffer 2.
&dataLen[1], // Buffer 2 length.
0 // Flags.
);
if(FAILED(hr))
return 0;
// Hopefully we didn't get too much data back!
if((dataLen[0]+dataLen[1]) > nSamplesWanted)
{
assert(false);
m_pCaptureBuffer->Unlock(pData[0], dataLen[0], pData[1], dataLen[1]);
return 0;
}
// Copy the data to the output.
memcpy(pOut, pData[0], dataLen[0]);
memcpy(&pOut[dataLen[0]/2], pData[1], dataLen[1]);
m_pCaptureBuffer->Unlock(pData[0], dataLen[0], pData[1], dataLen[1]);
// Update the shiz.
m_LastReadPos = dwReadPos;
return (dataLen[0] + dataLen[1]) >> 1;
}
int audioStreamer_ds::Read(char *buf, int len) // returns 0 if blocked, < 0 if error, > 0 if data
{
if (!m_inbuf) return -1;
if (!m_has_started)
{
m_inbuf->Start(DSCBSTART_LOOPING);
m_has_started=1;
}
int cappos;
m_inbuf->GetCurrentPosition(NULL,(DWORD *)&cappos);
if (cappos < m_last_pos) m_i_dw++;
m_last_pos=cappos;
if ((m_i_dw - m_i_lw) * m_totalbufsize + cappos - m_bufpos >= (unsigned int)m_totalbufsize/2) // detect overrun, set to cappos
{
m_i_lw=m_i_dw;
m_bufpos=0;
while (m_bufpos < cappos-m_bufsize) m_bufpos += m_bufsize;
if (m_bufpos >= m_totalbufsize)
{
m_i_lw++;
m_bufpos -= m_totalbufsize;
}
// audiostream_onover();
}
for (;;)
{
if (m_i_lw < m_i_dw || m_bufpos + m_bufsize < cappos) break;
Sleep(DS_SLEEP);
m_inbuf->GetCurrentPosition(NULL,(DWORD*)&cappos);
if (cappos < m_last_pos) m_i_dw++;
m_last_pos=cappos;
}
//audiostream_instance->g_sound_in_overruns = (m_i_lw < m_i_dw ? (m_totalbufsize+cappos) : cappos ) -m_bufpos;
void *v1=0, *v2=0;
DWORD lv1=0, lv2=0;
if (m_inbuf->Lock(m_bufpos,len,&v1,&lv1,&v2,&lv2,FALSE) == DS_OK)
{
int l1=min((int)lv1,len);
memcpy(buf,v1,l1);
if (l1 < len && v2 && lv2) memcpy(buf+l1,v2,min((int)lv2,len-l1));
m_inbuf->Unlock(v1,lv1,v2,lv2);
m_bufpos += len;
if (m_bufpos >= m_totalbufsize)
{
m_i_lw++;
m_bufpos -= m_totalbufsize;
}
}
else
{
return -1;
}
return len;
}