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Added basic sample rate control in the DirectSound module, removed chunk and buffer settings as they must be computed from rate. 

git-svn-id: http://voip.null.ro/svn/yate@3164 acf43c95-373e-0410-b603-e72c3f656dc1
This commit is contained in:
paulc 2010-03-31 13:53:44 +00:00
parent 42d4dded8f
commit 1bc096fa99
2 changed files with 67 additions and 94 deletions
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16
conf.d/dsoundchan.conf.sample
View File

@ -4,17 +4,5 @@
; primary: boolean: Use the primary playback sound buffer instead of a secondary ; primary: boolean: Use the primary playback sound buffer instead of a secondary
;primary=yes ;primary=yes
; There is not much tweaking possible with these settings. ; rate: int: Default sampling rate used, can be 8000, 16000, 32000
; Some sanity is enforced in code but actual working limits depend on the sound drivers ;rate=8000
; chunk: int: Number of bytes in the chunk of samples transferred at once
;chunk=320
; minsize: int: Number of bytes in buffer before we start playing back
;minsize=640
; bufsize: int: Bytes allocated for secondary playback and the record buffers
;bufsize=1280
; maxsize: int: Number of buffered bytes when we start dropping chunks
;maxsize=1600

145
modules/client/dsoundchan.cpp
View File

@ -43,18 +43,14 @@ namespace { // anonymous
// we should use the primary sound buffer else we will lose sound while we have no input focus // we should use the primary sound buffer else we will lose sound while we have no input focus
static bool s_primary = true; static bool s_primary = true;
// 20ms minimum chunk, 100ms buffer // default sampling rate
#define CHUNK_SIZE 320 static int s_rate = 8000;
static unsigned int s_chunk = CHUNK_SIZE;
static unsigned int s_minsize = 2*CHUNK_SIZE;
static unsigned int s_bufsize = 4*CHUNK_SIZE;
static unsigned int s_maxsize = 5*CHUNK_SIZE;
class DSoundSource : public DataSource class DSoundSource : public DataSource
{ {
friend class DSoundRec; friend class DSoundRec;
public: public:
DSoundSource(); DSoundSource(int rate = 8000);
~DSoundSource(); ~DSoundSource();
bool control(NamedList& msg); bool control(NamedList& msg);
private: private:
@ -65,7 +61,7 @@ class DSoundConsumer : public DataConsumer
{ {
friend class DSoundPlay; friend class DSoundPlay;
public: public:
DSoundConsumer(bool stereo = false); DSoundConsumer(int rate = 8000, bool stereo = false);
~DSoundConsumer(); ~DSoundConsumer();
virtual unsigned long Consume(const DataBlock &data, unsigned long tStamp, unsigned long flags); virtual unsigned long Consume(const DataBlock &data, unsigned long tStamp, unsigned long flags);
bool control(NamedList& msg); bool control(NamedList& msg);
@ -78,7 +74,7 @@ private:
class DSoundPlay : public Thread, public Mutex class DSoundPlay : public Thread, public Mutex
{ {
public: public:
DSoundPlay(DSoundConsumer* owner, LPGUID device = 0); DSoundPlay(DSoundConsumer* owner, DWORD rate, LPGUID device = 0);
virtual ~DSoundPlay(); virtual ~DSoundPlay();
virtual void run(); virtual void run();
virtual void cleanup(); virtual void cleanup();
@ -93,10 +89,12 @@ public:
bool control(NamedList& msg); bool control(NamedList& msg);
private: private:
DSoundConsumer* m_owner; DSoundConsumer* m_owner;
DWORD m_rate;
LPGUID m_device; LPGUID m_device;
LPDIRECTSOUND m_ds; LPDIRECTSOUND m_ds;
LPDIRECTSOUNDBUFFER m_dsb; LPDIRECTSOUNDBUFFER m_dsb;
DWORD m_buffSize; DWORD m_buffSize;
DWORD m_chunk;
DataBlock m_buf; DataBlock m_buf;
u_int64_t m_start; u_int64_t m_start;
u_int64_t m_total; u_int64_t m_total;
@ -106,7 +104,7 @@ private:
class DSoundRec : public Thread class DSoundRec : public Thread
{ {
public: public:
DSoundRec(DSoundSource* owner, LPGUID device = 0); DSoundRec(DSoundSource* owner, DWORD rate, LPGUID device = 0);
virtual ~DSoundRec(); virtual ~DSoundRec();
virtual void run(); virtual void run();
virtual void cleanup(); virtual void cleanup();
@ -120,6 +118,7 @@ public:
bool control(NamedList& msg); bool control(NamedList& msg);
private: private:
DSoundSource* m_owner; DSoundSource* m_owner;
DWORD m_rate;
LPGUID m_device; LPGUID m_device;
LPDIRECTSOUNDCAPTURE m_ds; LPDIRECTSOUNDCAPTURE m_ds;
LPDIRECTSOUNDCAPTUREBUFFER m_dsb; LPDIRECTSOUNDCAPTUREBUFFER m_dsb;
@ -133,7 +132,7 @@ private:
class DSoundChan : public Channel class DSoundChan : public Channel
{ {
public: public:
DSoundChan(); DSoundChan(int rate = 8000);
virtual ~DSoundChan(); virtual ~DSoundChan();
}; };
@ -162,10 +161,10 @@ private:
INIT_PLUGIN(SoundDriver); INIT_PLUGIN(SoundDriver);
DSoundPlay::DSoundPlay(DSoundConsumer* owner, LPGUID device) DSoundPlay::DSoundPlay(DSoundConsumer* owner, DWORD rate, LPGUID device)
: Thread("DirectSound Play",High), Mutex(false,"DSoundPlay"), : Thread("DirectSound Play",High), Mutex(false,"DSoundPlay"),
m_owner(0), m_device(device), m_ds(0), m_dsb(0), m_owner(0), m_rate(rate), m_device(device), m_ds(0), m_dsb(0),
m_buffSize(0), m_start(0), m_total(0) m_buffSize(0), m_chunk(320), m_start(0), m_total(0)
{ {
if (owner && owner->ref()) if (owner && owner->ref())
m_owner = owner; m_owner = owner;
@ -205,18 +204,19 @@ bool DSoundPlay::init()
// Set channel number depending data // Set channel number depending data
WORD nChannels = 1; WORD nChannels = 1;
DWORD nAvgBytesPerSec = 16000; // nSamplesPerSec * nBlockAlign. DWORD nAvgBytesPerSec = 2 * m_rate; // nSamplesPerSec * nBlockAlign.
WORD nBlockAlign = 2; // nChannels * wBitsPerSample / 8 WORD nBlockAlign = 2; // nChannels * wBitsPerSample / 8
if (m_owner && m_owner->m_stereo) { if (m_owner && m_owner->m_stereo) {
nChannels = 2; nChannels = 2;
nAvgBytesPerSec = 32000; nAvgBytesPerSec = 4 * m_rate;
nBlockAlign = 4; nBlockAlign = 4;
} }
m_chunk = nChannels * m_rate / 25; // 20ms of 2-byte samples
WAVEFORMATEX fmt; WAVEFORMATEX fmt;
fmt.wFormatTag = WAVE_FORMAT_PCM; fmt.wFormatTag = WAVE_FORMAT_PCM;
fmt.nChannels = nChannels; fmt.nChannels = nChannels;
fmt.nSamplesPerSec = 8000; fmt.nSamplesPerSec = m_rate;
fmt.nAvgBytesPerSec = nAvgBytesPerSec; fmt.nAvgBytesPerSec = nAvgBytesPerSec;
fmt.nBlockAlign = nBlockAlign; fmt.nBlockAlign = nBlockAlign;
fmt.wBitsPerSample = 16; fmt.wBitsPerSample = 16;
@ -230,7 +230,7 @@ bool DSoundPlay::init()
else { else {
bdesc.dwFlags |= DSBCAPS_GLOBALFOCUS; bdesc.dwFlags |= DSBCAPS_GLOBALFOCUS;
// we have to set format when creating secondary buffers // we have to set format when creating secondary buffers
bdesc.dwBufferBytes = s_bufsize; bdesc.dwBufferBytes = 4*m_chunk;
bdesc.lpwfxFormat = &fmt; bdesc.lpwfxFormat = &fmt;
} }
if (FAILED(hr = m_ds->CreateSoundBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) { if (FAILED(hr = m_ds->CreateSoundBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) {
@ -248,10 +248,10 @@ bool DSoundPlay::init()
} }
if ((fmt.wFormatTag != WAVE_FORMAT_PCM) || if ((fmt.wFormatTag != WAVE_FORMAT_PCM) ||
(fmt.nChannels != nChannels) || (fmt.nChannels != nChannels) ||
(fmt.nSamplesPerSec != 8000) || (fmt.nSamplesPerSec != m_rate) ||
(fmt.wBitsPerSample != 16)) { (fmt.wBitsPerSample != 16)) {
Debug(DebugGoOn,"DirectSound does not support 8000Hz 16bit %s PCM format, " Debug(DebugGoOn,"DirectSound does not support %dHz 16bit %s PCM format, "
"got fmt=%u, chans=%d samp=%d size=%u",nChannels == 1 ? "mono" : "stereo", "got fmt=%u, chans=%d samp=%d size=%u",m_rate,nChannels == 1 ? "mono" : "stereo",
fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample); fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample);
return false; return false;
} }
@ -283,12 +283,12 @@ void DSoundPlay::run()
else else
return; return;
DWORD writeOffs = 0; DWORD writeOffs = 0;
DWORD margin = s_chunk/4; DWORD margin = m_chunk/4;
bool first = true; bool first = true;
while (m_owner->refcount() > 1) { while (m_owner->refcount() > 1) {
msleep(1,true); msleep(1,true);
if (first) { if (first) {
if ((m_buf.length() < s_minsize) || !m_dsb) if ((m_buf.length() < 2*m_chunk) || !m_dsb)
continue; continue;
first = false; first = false;
m_dsb->GetCurrentPosition(NULL,&writeOffs); m_dsb->GetCurrentPosition(NULL,&writeOffs);
@ -316,10 +316,10 @@ void DSoundPlay::run()
writeOffs,adjOffs,playPos,writePos); writeOffs,adjOffs,playPos,writePos);
writeOffs = adjOffs; writeOffs = adjOffs;
} }
bool hasData = (m_buf.length() >= s_chunk); bool hasData = (m_buf.length() >= m_chunk);
if (!(adjust || hasData)) { if (!(adjust || hasData)) {
// don't fill the buffer if we still have at least one chunk until underflow // don't fill the buffer if we still have at least one chunk until underflow
if ((m_buffSize + writeOffs - writePos) % m_buffSize >= s_chunk) if ((m_buffSize + writeOffs - writePos) % m_buffSize >= m_chunk)
break; break;
} }
void* buf = 0; void* buf = 0;
@ -327,13 +327,13 @@ void DSoundPlay::run()
DWORD len = 0; DWORD len = 0;
DWORD len2 = 0; DWORD len2 = 0;
// locking will prevent us to skip ahead and overwrite the play position // locking will prevent us to skip ahead and overwrite the play position
HRESULT hr = m_dsb->Lock(writeOffs,s_chunk,&buf,&len,&buf2,&len2,0); HRESULT hr = m_dsb->Lock(writeOffs,m_chunk,&buf,&len,&buf2,&len2,0);
if (FAILED(hr)) { if (FAILED(hr)) {
writeOffs = 0; writeOffs = 0;
if ((hr == DSERR_BUFFERLOST) && SUCCEEDED(m_dsb->Restore())) { if ((hr == DSERR_BUFFERLOST) && SUCCEEDED(m_dsb->Restore())) {
m_dsb->Play(0,0,DSBPLAY_LOOPING); m_dsb->Play(0,0,DSBPLAY_LOOPING);
m_dsb->GetCurrentPosition(NULL,&writeOffs); m_dsb->GetCurrentPosition(NULL,&writeOffs);
writeOffs = (s_chunk/4 + writeOffs) % m_buffSize; writeOffs = (margin + writeOffs) % m_buffSize;
Debug(&__plugin,DebugAll,"DirectSound buffer lost and restored, playing at %u", Debug(&__plugin,DebugAll,"DirectSound buffer lost and restored, playing at %u",
writeOffs); writeOffs);
} }
@ -356,15 +356,15 @@ void DSoundPlay::run()
::memset(buf2,0,len2); ::memset(buf2,0,len2);
} }
m_dsb->Unlock(buf,len,buf2,len2); m_dsb->Unlock(buf,len,buf2,len2);
m_total += s_chunk; m_total += m_chunk;
m_buf.cut(-(int)s_chunk); m_buf.cut(-(int)m_chunk);
unlock(); unlock();
#ifdef DEBUG #ifdef DEBUG
if (!hasData) if (!hasData)
Debug(&__plugin,DebugInfo,"Underflow, filled %u bytes at %u, p=%u w=%u", Debug(&__plugin,DebugInfo,"Underflow, filled %u bytes at %u, p=%u w=%u",
s_chunk,writeOffs,playPos,writePos); m_chunk,writeOffs,playPos,writePos);
#endif #endif
writeOffs += s_chunk; writeOffs += m_chunk;
if (writeOffs >= m_buffSize) if (writeOffs >= m_buffSize)
writeOffs -= m_buffSize; writeOffs -= m_buffSize;
XDebug(&__plugin,DebugAll,"Locked %p,%d %p,%d",buf,len,buf2,len2); XDebug(&__plugin,DebugAll,"Locked %p,%d %p,%d",buf,len,buf2,len2);
@ -420,7 +420,7 @@ void DSoundPlay::put(const DataBlock& data)
if (!m_dsb) if (!m_dsb)
return; return;
lock(); lock();
if (m_buf.length() + data.length() <= s_maxsize) if (m_buf.length() + data.length() <= m_buffSize + m_chunk)
m_buf += data; m_buf += data;
else else
Debug(&__plugin,DebugMild,"DSoundPlay skipped %u bytes, buffer is full",data.length()); Debug(&__plugin,DebugMild,"DSoundPlay skipped %u bytes, buffer is full",data.length());
@ -428,9 +428,9 @@ void DSoundPlay::put(const DataBlock& data)
} }
DSoundRec::DSoundRec(DSoundSource* owner, LPGUID device) DSoundRec::DSoundRec(DSoundSource* owner, DWORD rate, LPGUID device)
: Thread("DirectSound Rec",High), : Thread("DirectSound Rec",High),
m_owner(0), m_device(device), m_ds(0), m_dsb(0), m_owner(0), m_rate(rate), m_device(device), m_ds(0), m_dsb(0),
m_buffSize(0), m_readPos(0), m_start(0), m_total(0), m_rshift(0) m_buffSize(0), m_readPos(0), m_start(0), m_total(0), m_rshift(0)
{ {
if (owner && owner->ref()) if (owner && owner->ref())
@ -464,8 +464,8 @@ bool DSoundRec::init()
WAVEFORMATEX fmt; WAVEFORMATEX fmt;
fmt.wFormatTag = WAVE_FORMAT_PCM; fmt.wFormatTag = WAVE_FORMAT_PCM;
fmt.nChannels = 1; fmt.nChannels = 1;
fmt.nSamplesPerSec = 8000; fmt.nSamplesPerSec = m_rate;
fmt.nAvgBytesPerSec = 16000; fmt.nAvgBytesPerSec = 2 * m_rate;
fmt.nBlockAlign = 2; fmt.nBlockAlign = 2;
fmt.wBitsPerSample = 16; fmt.wBitsPerSample = 16;
fmt.cbSize = 0; fmt.cbSize = 0;
@ -473,7 +473,7 @@ bool DSoundRec::init()
ZeroMemory(&bdesc, sizeof(bdesc)); ZeroMemory(&bdesc, sizeof(bdesc));
bdesc.dwSize = sizeof(bdesc); bdesc.dwSize = sizeof(bdesc);
bdesc.dwFlags = DSCBCAPS_WAVEMAPPED; bdesc.dwFlags = DSCBCAPS_WAVEMAPPED;
bdesc.dwBufferBytes = s_bufsize; bdesc.dwBufferBytes = 4 * m_rate / 25;
bdesc.lpwfxFormat = &fmt; bdesc.lpwfxFormat = &fmt;
if (FAILED(hr = m_ds->CreateCaptureBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) { if (FAILED(hr = m_ds->CreateCaptureBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) {
Debug(DebugGoOn,"Could not create the DirectSoundCapture buffer, code 0x%X",hr); Debug(DebugGoOn,"Could not create the DirectSoundCapture buffer, code 0x%X",hr);
@ -485,10 +485,10 @@ bool DSoundRec::init()
} }
if ((fmt.wFormatTag != WAVE_FORMAT_PCM) || if ((fmt.wFormatTag != WAVE_FORMAT_PCM) ||
(fmt.nChannels != 1) || (fmt.nChannels != 1) ||
(fmt.nSamplesPerSec != 8000) || (fmt.nSamplesPerSec != m_rate) ||
(fmt.wBitsPerSample != 16)) { (fmt.wBitsPerSample != 16)) {
Debug(DebugGoOn,"DirectSoundCapture does not support 8000Hz 16bit mono PCM format, " Debug(DebugGoOn,"DirectSoundCapture does not support %dHz 16bit mono PCM format, "
"got fmt=%u, chans=%d samp=%d size=%u", "got fmt=%u, chans=%d samp=%d size=%u",m_rate,
fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample); fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample);
return false; return false;
} }
@ -512,6 +512,7 @@ void DSoundRec::run()
if (!init()) if (!init())
return; return;
Debug(&__plugin,DebugInfo,"DSoundRec is initialized and running"); Debug(&__plugin,DebugInfo,"DSoundRec is initialized and running");
DWORD chunk = m_rate / 25; // 20ms of 2-byte samples
m_start = Time::now(); m_start = Time::now();
if (m_owner) if (m_owner)
m_owner->m_dsound = this; m_owner->m_dsound = this;
@ -526,13 +527,13 @@ void DSoundRec::run()
if (pos < m_readPos) if (pos < m_readPos)
pos += m_buffSize; pos += m_buffSize;
pos -= m_readPos; pos -= m_readPos;
if (pos < s_chunk) if (pos < chunk)
continue; continue;
void* buf = 0; void* buf = 0;
void* buf2 = 0; void* buf2 = 0;
DWORD len = 0; DWORD len = 0;
DWORD len2 = 0; DWORD len2 = 0;
if (FAILED(m_dsb->Lock(m_readPos,s_chunk,&buf,&len,&buf2,&len2,0))) if (FAILED(m_dsb->Lock(m_readPos,chunk,&buf,&len,&buf2,&len2,0)))
continue; continue;
DataBlock data(0,len+len2); DataBlock data(0,len+len2);
::memcpy(data.data(),buf,len); ::memcpy(data.data(),buf,len);
@ -592,10 +593,12 @@ bool DSoundRec::control(TelEngine::NamedList &msg)
} }
DSoundSource::DSoundSource() DSoundSource::DSoundSource(int rate)
: m_dsound(0) : m_dsound(0)
{ {
DSoundRec* ds = new DSoundRec(this); if (rate != 8000)
m_format << "/" << rate;
DSoundRec* ds = new DSoundRec(this,rate);
ds->startup(); ds->startup();
} }
@ -613,11 +616,13 @@ bool DSoundSource::control(NamedList& msg)
} }
DSoundConsumer::DSoundConsumer(bool stereo) DSoundConsumer::DSoundConsumer(int rate, bool stereo)
: DataConsumer(stereo ? "2*slin" : "slin"), : DataConsumer(stereo ? "2*slin" : "slin"),
m_dsound(0), m_stereo(stereo) m_dsound(0), m_stereo(stereo)
{ {
DSoundPlay* ds = new DSoundPlay(this); if (rate != 8000)
m_format << "/" << rate;
DSoundPlay* ds = new DSoundPlay(this,rate);
ds->startup(); ds->startup();
} }
@ -644,15 +649,15 @@ bool DSoundConsumer::control(NamedList& msg)
} }
DSoundChan::DSoundChan() DSoundChan::DSoundChan(int rate)
: Channel(__plugin) : Channel(__plugin)
{ {
Debug(this,DebugAll,"DSoundChan::DSoundChan() [%p]",this); Debug(this,DebugAll,"DSoundChan::DSoundChan(%d) [%p]",rate,this);
setConsumer(new DSoundConsumer); setConsumer(new DSoundConsumer(rate));
getConsumer()->deref(); getConsumer()->deref();
Thread::msleep(50); Thread::msleep(50);
setSource(new DSoundSource); setSource(new DSoundSource(rate));
getSource()->deref(); getSource()->deref();
Thread::msleep(50); Thread::msleep(50);
} }
@ -693,15 +698,17 @@ bool AttachHandler::received(Message &msg)
return false; return false;
} }
int rate = msg.getIntValue("rate",s_rate);
if (cons) { if (cons) {
DSoundConsumer* c = new DSoundConsumer(msg.getBoolValue("stereo")); DSoundConsumer* c = new DSoundConsumer(rate,msg.getBoolValue("stereo"));
dd->setConsumer(c); dd->setConsumer(c);
c->deref(); c->deref();
Thread::msleep(50); Thread::msleep(50);
} }
if (src) { if (src) {
DSoundSource* s = new DSoundSource; DSoundSource* s = new DSoundSource(rate);
dd->setSource(s); dd->setSource(s);
s->deref(); s->deref();
Thread::msleep(50); Thread::msleep(50);
@ -717,7 +724,7 @@ bool SoundDriver::msgExecute(Message& msg, String& dest)
{ {
CallEndpoint* ch = static_cast<CallEndpoint*>(msg.userData()); CallEndpoint* ch = static_cast<CallEndpoint*>(msg.userData());
if (ch) { if (ch) {
DSoundChan *ds = new DSoundChan; DSoundChan *ds = new DSoundChan(msg.getIntValue("rate",s_rate));
if (ch->connect(ds,msg.getValue("reason"))) { if (ch->connect(ds,msg.getValue("reason"))) {
msg.setParam("peerid",ds->id()); msg.setParam("peerid",ds->id());
ds->deref(); ds->deref();
@ -751,7 +758,7 @@ bool SoundDriver::msgExecute(Message& msg, String& dest)
} }
m = "call.execute"; m = "call.execute";
m.addParam("callto",callto); m.addParam("callto",callto);
DSoundChan *ds = new DSoundChan; DSoundChan *ds = new DSoundChan(msg.getIntValue("rate",8000));
m.setParam("targetid",ds->id()); m.setParam("targetid",ds->id());
m.userData(ds); m.userData(ds);
if (Engine::dispatch(m)) { if (Engine::dispatch(m)) {
@ -783,33 +790,11 @@ void SoundDriver::initialize()
Output("Initializing module DirectSound"); Output("Initializing module DirectSound");
setup(0,true); // no need to install notifications setup(0,true); // no need to install notifications
Driver::initialize(); Driver::initialize();
Configuration cfg(Engine::configFile("dsoundchan"));
s_rate = cfg.getIntValue("general","rate",8000);
// prefer primary buffer as we try to retain control of audio board
s_primary = cfg.getBoolValue("general","primary",true);
if (!m_handler) { if (!m_handler) {
Configuration cfg(Engine::configFile("dsoundchan"));
s_chunk = cfg.getIntValue("general","chunk",CHUNK_SIZE);
// make sure the chunk is even sized and has some decent limits (20-50 ms)
s_chunk &= ~1;
if (s_chunk < 320)
s_chunk = 320;
if (s_chunk > 800)
s_chunk = 800;
s_minsize = cfg.getIntValue("general","minsize",2*s_chunk);
s_bufsize = cfg.getIntValue("general","bufsize",4*s_chunk);
s_maxsize = cfg.getIntValue("general","maxsize",5*s_chunk);
// the buffer MUST hold at least one chunk and about 15ms of audio - we allow 30
if (s_bufsize < s_chunk + 480)
s_bufsize = s_chunk + 480;
// also keep it under 2s and even sized
if (s_bufsize > 32000)
s_bufsize = 32000;
s_bufsize &= ~1;
// make sure playback can ever start
if (s_minsize > s_bufsize - s_chunk)
s_minsize = s_bufsize - s_chunk;
// and that we don't do stupid drops
if (s_maxsize < s_bufsize + s_chunk)
s_maxsize = s_bufsize + s_chunk;
// prefer primary buffer as we try to retain control of audio board
s_primary = cfg.getBoolValue("general","primary",true);
m_handler = new AttachHandler; m_handler = new AttachHandler;
Engine::install(m_handler); Engine::install(m_handler);
} }