yate/modules/client/dsoundchan.cpp

811 lines
22 KiB
C++

/**
* dsoundchan.cpp
* This file is part of the YATE Project http://YATE.null.ro
*
* DirectSound channel driver for Windows.
*
* Yet Another Telephony Engine - a fully featured software PBX and IVR
* Copyright (C) 2004-2014 Null Team
*
* This software is distributed under multiple licenses;
* see the COPYING file in the main directory for licensing
* information for this specific distribution.
*
* This use of this software may be subject to additional restrictions.
* See the LEGAL file in the main directory for details.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.
*/
// define DCOM before including windows.h so advanced COM functions can be compiled
#define _WIN32_DCOM
#include <yatephone.h>
#ifndef _WINDOWS
#error This module is only for Windows
#else
#include <string.h>
// initialize the GUIDs so we don't need to link against dsound.lib
#include <initguid.h>
#include <dsound.h>
using namespace TelEngine;
namespace { // anonymous
// we should use the primary sound buffer else we will lose sound while we have no input focus
static bool s_primary = true;
// default sampling rate
static int s_rate = 8000;
class DSoundSource : public DataSource
{
friend class DSoundRec;
public:
DSoundSource(int rate = 8000);
~DSoundSource();
bool control(NamedList& msg);
private:
DSoundRec* m_dsound;
};
class DSoundConsumer : public DataConsumer
{
friend class DSoundPlay;
public:
DSoundConsumer(int rate = 8000, bool stereo = false);
~DSoundConsumer();
virtual unsigned long Consume(const DataBlock &data, unsigned long tStamp, unsigned long flags);
bool control(NamedList& msg);
private:
DSoundPlay* m_dsound;
bool m_stereo;
};
// all DirectSound play related objects are created in this thread's apartment
class DSoundPlay : public Thread, public Mutex
{
public:
DSoundPlay(DSoundConsumer* owner, DWORD rate, LPGUID device = 0);
virtual ~DSoundPlay();
virtual void run();
virtual void cleanup();
bool init();
inline void terminate()
{ m_owner = 0; }
inline LPDIRECTSOUND dsound() const
{ return m_ds; }
inline LPDIRECTSOUNDBUFFER buffer() const
{ return m_dsb; }
void put(const DataBlock& data);
bool control(NamedList& msg);
private:
DSoundConsumer* m_owner;
DWORD m_rate;
LPGUID m_device;
LPDIRECTSOUND m_ds;
LPDIRECTSOUNDBUFFER m_dsb;
DWORD m_buffSize;
DWORD m_chunk;
DataBlock m_buf;
u_int64_t m_start;
u_int64_t m_total;
};
// all DirectSound record related objects are created in this thread's apartment
class DSoundRec : public Thread
{
public:
DSoundRec(DSoundSource* owner, DWORD rate, LPGUID device = 0);
virtual ~DSoundRec();
virtual void run();
virtual void cleanup();
bool init();
inline void terminate()
{ m_owner = 0; Thread::msleep(10); }
inline LPDIRECTSOUNDCAPTURE dsound() const
{ return m_ds; }
inline LPDIRECTSOUNDCAPTUREBUFFER buffer() const
{ return m_dsb; }
bool control(NamedList& msg);
private:
DSoundSource* m_owner;
DWORD m_rate;
LPGUID m_device;
LPDIRECTSOUNDCAPTURE m_ds;
LPDIRECTSOUNDCAPTUREBUFFER m_dsb;
DWORD m_buffSize;
DWORD m_readPos;
u_int64_t m_start;
u_int64_t m_total;
int m_rshift;
};
class DSoundChan : public Channel
{
public:
DSoundChan(int rate = 8000);
virtual ~DSoundChan();
};
class AttachHandler;
class SoundDriver : public Driver
{
friend class DSoundPlay;
friend class DSoundRec;
public:
SoundDriver();
~SoundDriver();
virtual void initialize();
virtual bool msgExecute(Message& msg, String& dest);
private:
AttachHandler* m_handler;
};
INIT_PLUGIN(SoundDriver);
class AttachHandler : public MessageHandler
{
public:
AttachHandler()
: MessageHandler("chan.attach",100,__plugin.name())
{ }
virtual bool received(Message &msg);
};
DSoundPlay::DSoundPlay(DSoundConsumer* owner, DWORD rate, LPGUID device)
: Thread("DirectSound Play",High), Mutex(false,"DSoundPlay"),
m_owner(0), m_rate(rate), m_device(device), m_ds(0), m_dsb(0),
m_buffSize(0), m_chunk(320), m_start(0), m_total(0)
{
if (owner && owner->ref())
m_owner = owner;
}
DSoundPlay::~DSoundPlay()
{
if (m_start && m_total) {
unsigned int rate = (unsigned int)(m_total * 1000000 / (Time::now() - m_start));
Debug(&__plugin,DebugInfo,"DSoundPlay transferred %u bytes/s, total " FMT64U,rate,m_total);
}
}
bool DSoundPlay::init()
{
HRESULT hr;
if (FAILED(hr = ::CoInitializeEx(NULL,COINIT_MULTITHREADED))) {
Debug(DebugCrit,"Could not initialize the COM library, code 0x%X",hr);
return false;
}
if (FAILED(hr = ::CoCreateInstance(CLSID_DirectSound, NULL, CLSCTX_INPROC_SERVER,
IID_IDirectSound, (void**)&m_ds)) || !m_ds) {
Debug(DebugCrit,"Could not create the DirectSound object, code 0x%X",hr);
return false;
}
if (FAILED(hr = m_ds->Initialize(m_device))) {
Debug(DebugWarn,"Could not initialize the DirectSound object, code 0x%X",hr);
return false;
}
HWND wnd = GetForegroundWindow();
if (!wnd)
wnd = GetDesktopWindow();
if (FAILED(hr = m_ds->SetCooperativeLevel(wnd,s_primary ? DSSCL_WRITEPRIMARY : DSSCL_EXCLUSIVE))) {
Debug(DebugCrit,"Could not set the DirectSound cooperative level, code 0x%X",hr);
return false;
}
// Set channel number depending data
WORD nChannels = 1;
DWORD nAvgBytesPerSec = 2 * m_rate; // nSamplesPerSec * nBlockAlign.
WORD nBlockAlign = 2; // nChannels * wBitsPerSample / 8
if (m_owner && m_owner->m_stereo) {
nChannels = 2;
nAvgBytesPerSec = 4 * m_rate;
nBlockAlign = 4;
}
m_chunk = nChannels * m_rate / 25; // 20ms of 2-byte samples
WAVEFORMATEX fmt;
fmt.wFormatTag = WAVE_FORMAT_PCM;
fmt.nChannels = nChannels;
fmt.nSamplesPerSec = m_rate;
fmt.nAvgBytesPerSec = nAvgBytesPerSec;
fmt.nBlockAlign = nBlockAlign;
fmt.wBitsPerSample = 16;
fmt.cbSize = 0;
DSBUFFERDESC bdesc;
ZeroMemory(&bdesc, sizeof(bdesc));
bdesc.dwSize = sizeof(bdesc);
bdesc.dwFlags = DSBCAPS_CTRLVOLUME;
if (s_primary)
bdesc.dwFlags |= DSBCAPS_PRIMARYBUFFER | DSBCAPS_STICKYFOCUS;
else {
bdesc.dwFlags |= DSBCAPS_GLOBALFOCUS;
// we have to set format when creating secondary buffers
bdesc.dwBufferBytes = 4*m_chunk;
bdesc.lpwfxFormat = &fmt;
}
if (FAILED(hr = m_ds->CreateSoundBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) {
Debug(DebugCrit,"Could not create the DirectSound buffer, code 0x%X",hr);
return false;
}
// format can be changed only for primary buffers
if (s_primary && FAILED(hr = m_dsb->SetFormat(&fmt))) {
Debug(DebugWarn,"Could not set the DirectSound buffer format, code 0x%X",hr);
return false;
}
if (FAILED(hr = m_dsb->GetFormat(&fmt,sizeof(fmt),0))) {
Debug(DebugWarn,"Could not get the DirectSound buffer format, code 0x%X",hr);
return false;
}
if ((fmt.wFormatTag != WAVE_FORMAT_PCM) ||
(fmt.nChannels != nChannels) ||
(fmt.nSamplesPerSec != m_rate) ||
(fmt.wBitsPerSample != 16)) {
Debug(DebugWarn,"DirectSound does not support %dHz 16bit %s PCM format, "
"got fmt=%u, chans=%d samp=%d size=%u",m_rate,nChannels == 1 ? "mono" : "stereo",
fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample);
return false;
}
DSBCAPS caps;
caps.dwSize = sizeof(caps);
if (FAILED(hr = m_dsb->GetCaps(&caps))) {
Debug(DebugCrit,"Could not get the DirectSound buffer capabilities, code 0x%X",hr);
return false;
}
m_buffSize = caps.dwBufferBytes;
Debug(&__plugin,DebugInfo,"DirectSound buffer size %u",m_buffSize);
if (FAILED(hr = m_dsb->Play(0,0,DSBPLAY_LOOPING))) {
if ((hr != DSERR_BUFFERLOST) || FAILED(hr = m_dsb->Restore())) {
Debug(DebugWarn,"Could not play the DirectSound buffer, code 0x%X",hr);
return false;
}
m_dsb->Play(0,0,DSBPLAY_LOOPING);
}
return true;
}
void DSoundPlay::run()
{
if (!init())
return;
Debug(&__plugin,DebugInfo,"DSoundPlay is initialized and running");
if (m_owner)
m_owner->m_dsound = this;
else
return;
DWORD writeOffs = 0;
DWORD margin = m_chunk/4;
bool first = true;
while (m_owner->refcount() > 1) {
msleep(1,true);
if (first) {
if ((m_buf.length() < 2*m_chunk) || !m_dsb)
continue;
first = false;
m_dsb->GetCurrentPosition(NULL,&writeOffs);
writeOffs = (margin + writeOffs) % m_buffSize;
Debug(&__plugin,DebugAll,"DSoundPlay has %u in buffer and starts playing at %u",
m_buf.length(),writeOffs);
m_start = Time::now();
}
while (m_dsb) {
DWORD playPos = 0;
DWORD writePos = 0;
bool adjust = false;
// check if we slipped behind and advance our pointer if so
if (SUCCEEDED(m_dsb->GetCurrentPosition(&playPos,&writePos))) {
if (playPos < writePos)
// not wrapped - have to adjust if our pointer falls between play and write
adjust = (playPos < writeOffs) && (writeOffs < writePos);
else
// only write offset has wrapped - adjust if we are outside
adjust = (writeOffs < writePos) || (playPos <= writeOffs) ;
}
if (adjust) {
DWORD adjOffs = (margin + writePos) % m_buffSize;
Debug(&__plugin,DebugNote,"Slip detected, changing write offs from %u to %u, p=%u w=%u",
writeOffs,adjOffs,playPos,writePos);
writeOffs = adjOffs;
}
bool hasData = (m_buf.length() >= m_chunk);
if (!(adjust || hasData)) {
// don't fill the buffer if we still have at least one chunk until underflow
if ((m_buffSize + writeOffs - writePos) % m_buffSize >= m_chunk)
break;
}
void* buf = 0;
void* buf2 = 0;
DWORD len = 0;
DWORD len2 = 0;
// locking will prevent us to skip ahead and overwrite the play position
HRESULT hr = m_dsb->Lock(writeOffs,m_chunk,&buf,&len,&buf2,&len2,0);
if (FAILED(hr)) {
writeOffs = 0;
if ((hr == DSERR_BUFFERLOST) && SUCCEEDED(m_dsb->Restore())) {
m_dsb->Play(0,0,DSBPLAY_LOOPING);
m_dsb->GetCurrentPosition(NULL,&writeOffs);
writeOffs = (margin + writeOffs) % m_buffSize;
Debug(&__plugin,DebugAll,"DirectSound buffer lost and restored, playing at %u",
writeOffs);
}
else {
lock();
m_buf.clear();
unlock();
}
continue;
}
lock();
if (hasData) {
::memcpy(buf,m_buf.data(),len);
if (buf2)
::memcpy(buf2,((const char*)m_buf.data())+len,len2);
}
else {
::memset(buf,0,len);
if (buf2)
::memset(buf2,0,len2);
}
m_dsb->Unlock(buf,len,buf2,len2);
m_total += m_chunk;
m_buf.cut(-(int)m_chunk);
unlock();
#ifdef DEBUG
if (!hasData)
Debug(&__plugin,DebugInfo,"Underflow, filled %u bytes at %u, p=%u w=%u",
m_chunk,writeOffs,playPos,writePos);
#endif
writeOffs += m_chunk;
if (writeOffs >= m_buffSize)
writeOffs -= m_buffSize;
XDebug(&__plugin,DebugAll,"Locked %p,%d %p,%d",buf,len,buf2,len2);
}
}
}
bool DSoundPlay::control(NamedList& msg)
{
bool ok = false;
LONG val = 0;
int outValue = msg.getIntValue("out_volume",-1);
HRESULT hr; // we need it for debugging
if ((outValue >= 0) && (outValue <= 100)) {
// convert 0...100 to 0...-50.00 dB
val = (outValue - 100) * 50;
ok = ((hr = m_dsb->SetVolume(val)) == DS_OK);
}
if ((hr = m_dsb->GetVolume(&val)) == DS_OK) {
// convert back 0...-50.0 dB to 0...100, watch out for values up to -100.00 dB
outValue = (5000 + val) / 50;
if (outValue < 0)
outValue = 0;
msg.setParam("out_volume", String(outValue));
}
return TelEngine::controlReturn(&msg,ok);
}
void DSoundPlay::cleanup()
{
Debug(DebugInfo,"DSoundPlay cleaning up");
if (m_owner) {
m_owner->m_dsound = 0;
if (m_owner->refcount() > 1)
Debug(&__plugin,DebugWarn,"DSoundPlay destroyed while consumer is still active");
TelEngine::destruct(m_owner);
}
if (m_dsb) {
m_dsb->Stop();
m_dsb->Release();
m_dsb = 0;
}
if (m_ds) {
m_ds->Release();
m_ds = 0;
}
::CoUninitialize();
}
void DSoundPlay::put(const DataBlock& data)
{
if (!m_dsb)
return;
lock();
if (m_buf.length() + data.length() <= m_buffSize + m_chunk)
m_buf += data;
else
Debug(&__plugin,DebugMild,"DSoundPlay skipped %u bytes, buffer is full",data.length());
unlock();
}
DSoundRec::DSoundRec(DSoundSource* owner, DWORD rate, LPGUID device)
: Thread("DirectSound Rec",High),
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)
{
if (owner && owner->ref())
m_owner = owner;
}
DSoundRec::~DSoundRec()
{
if (m_start && m_total) {
unsigned int rate = (unsigned int)(m_total * 1000000 / (Time::now() - m_start));
Debug(&__plugin,DebugInfo,"DSoundRec transferred %u bytes/s, total " FMT64U,rate,m_total);
}
}
bool DSoundRec::init()
{
HRESULT hr;
if (FAILED(hr = ::CoInitializeEx(NULL,COINIT_MULTITHREADED))) {
Debug(DebugCrit,"Could not initialize the COM library, code 0x%X",hr);
return false;
}
if (FAILED(hr = ::CoCreateInstance(CLSID_DirectSoundCapture, NULL, CLSCTX_INPROC_SERVER,
IID_IDirectSoundCapture, (void**)&m_ds)) || !m_ds) {
Debug(DebugCrit,"Could not create the DirectSoundCapture object, code 0x%X",hr);
return false;
}
if (FAILED(hr = m_ds->Initialize(m_device))) {
Debug(DebugWarn,"Could not initialize the DirectSoundCapture object, code 0x%X",hr);
return false;
}
WAVEFORMATEX fmt;
fmt.wFormatTag = WAVE_FORMAT_PCM;
fmt.nChannels = 1;
fmt.nSamplesPerSec = m_rate;
fmt.nAvgBytesPerSec = 2 * m_rate;
fmt.nBlockAlign = 2;
fmt.wBitsPerSample = 16;
fmt.cbSize = 0;
DSCBUFFERDESC bdesc;
ZeroMemory(&bdesc, sizeof(bdesc));
bdesc.dwSize = sizeof(bdesc);
bdesc.dwFlags = DSCBCAPS_WAVEMAPPED;
bdesc.dwBufferBytes = 4 * m_rate / 25;
bdesc.lpwfxFormat = &fmt;
if (FAILED(hr = m_ds->CreateCaptureBuffer(&bdesc, &m_dsb, NULL)) || !m_dsb) {
Debug(DebugCrit,"Could not create the DirectSoundCapture buffer, code 0x%X",hr);
return false;
}
if (FAILED(hr = m_dsb->GetFormat(&fmt,sizeof(fmt),0))) {
Debug(DebugCrit,"Could not get the DirectSoundCapture buffer format, code 0x%X",hr);
return false;
}
if ((fmt.wFormatTag != WAVE_FORMAT_PCM) ||
(fmt.nChannels != 1) ||
(fmt.nSamplesPerSec != m_rate) ||
(fmt.wBitsPerSample != 16)) {
Debug(DebugWarn,"DirectSoundCapture does not support %dHz 16bit mono PCM format, "
"got fmt=%u, chans=%d samp=%d size=%u",m_rate,
fmt.wFormatTag,fmt.nChannels,fmt.nSamplesPerSec,fmt.wBitsPerSample);
return false;
}
DSCBCAPS caps;
caps.dwSize = sizeof(caps);
if (FAILED(hr = m_dsb->GetCaps(&caps))) {
Debug(DebugCrit,"Could not get the DirectSoundCapture buffer capabilities, code 0x%X",hr);
return false;
}
m_buffSize = caps.dwBufferBytes;
Debug(&__plugin,DebugInfo,"DirectSoundCapture buffer size %u",m_buffSize);
if (FAILED(hr = m_dsb->Start(DSCBSTART_LOOPING))) {
Debug(DebugWarn,"Could not record to the DirectSoundCapture buffer, code 0x%X",hr);
return false;
}
return true;
}
void DSoundRec::run()
{
if (!init())
return;
Debug(&__plugin,DebugInfo,"DSoundRec is initialized and running");
DWORD chunk = m_rate / 25; // 20ms of 2-byte samples
m_start = Time::now();
if (m_owner)
m_owner->m_dsound = this;
else
return;
while (m_owner->refcount() > 1) {
msleep(1,true);
if (m_dsb) {
DWORD pos = 0;
if (FAILED(m_dsb->GetCurrentPosition(0,&pos)))
continue;
if (pos < m_readPos)
pos += m_buffSize;
pos -= m_readPos;
if (pos < chunk)
continue;
void* buf = 0;
void* buf2 = 0;
DWORD len = 0;
DWORD len2 = 0;
if (FAILED(m_dsb->Lock(m_readPos,chunk,&buf,&len,&buf2,&len2,0)))
continue;
DataBlock data(0,len+len2);
::memcpy(data.data(),buf,len);
if (buf2)
::memcpy(((char*)data.data())+len,buf2,len2);
m_dsb->Unlock(buf,len,buf2,len2);
m_total += (len+len2);
m_readPos += (len+len2);
if (m_readPos >= m_buffSize)
m_readPos -= m_buffSize;
if (m_rshift) {
// apply volume attenuation
signed short* s = (signed short*)data.data();
for (unsigned int n = data.length() / 2; n--; s++)
*s >>= m_rshift;
}
if (m_owner)
m_owner->Forward(data);
}
}
}
void DSoundRec::cleanup()
{
Debug(&__plugin,DebugInfo,"DSoundRec cleaning up");
if (m_owner) {
m_owner->m_dsound = 0;
if (m_owner->refcount() > 1)
Debug(&__plugin,DebugWarn,"DSoundRec destroyed while source is still active");
TelEngine::destruct(m_owner);
}
if (m_dsb) {
m_dsb->Stop();
m_dsb->Release();
m_dsb = 0;
}
if (m_ds) {
m_ds->Release();
m_ds = 0;
}
::CoUninitialize();
}
bool DSoundRec::control(TelEngine::NamedList &msg)
{
bool ok = false;
int inValue = msg.getIntValue("in_volume",-1);
if ((inValue >= 0) && (inValue <= 100)) {
// convert 0...100 to a 10...0 right shift count
m_rshift = (105 - inValue) / 10;
ok = true;
}
inValue = (10 - m_rshift) * 10;
msg.setParam("in_volume", String(inValue));
return TelEngine::controlReturn(&msg,ok);
}
DSoundSource::DSoundSource(int rate)
: m_dsound(0)
{
if (rate != 8000)
m_format << "/" << rate;
DSoundRec* ds = new DSoundRec(this,rate);
ds->startup();
}
DSoundSource::~DSoundSource()
{
if (m_dsound)
m_dsound->terminate();
}
bool DSoundSource::control(NamedList& msg)
{
if (m_dsound)
return m_dsound->control(msg);
return TelEngine::controlReturn(&msg,false);
}
DSoundConsumer::DSoundConsumer(int rate, bool stereo)
: DataConsumer(stereo ? "2*slin" : "slin"),
m_dsound(0), m_stereo(stereo)
{
if (rate != 8000)
m_format << "/" << rate;
DSoundPlay* ds = new DSoundPlay(this,rate);
ds->startup();
}
DSoundConsumer::~DSoundConsumer()
{
if (m_dsound)
m_dsound->terminate();
}
unsigned long DSoundConsumer::Consume(const DataBlock &data, unsigned long tStamp, unsigned long flags)
{
if (m_dsound) {
m_dsound->put(data);
return invalidStamp();
}
return 0;
}
bool DSoundConsumer::control(NamedList& msg)
{
if (m_dsound)
return m_dsound->control(msg);
return false;
}
DSoundChan::DSoundChan(int rate)
: Channel(__plugin)
{
Debug(this,DebugAll,"DSoundChan::DSoundChan(%d) [%p]",rate,this);
setConsumer(new DSoundConsumer(rate));
getConsumer()->deref();
Thread::msleep(50);
setSource(new DSoundSource(rate));
getSource()->deref();
Thread::msleep(50);
}
DSoundChan::~DSoundChan()
{
Debug(this,DebugAll,"DSoundChan::~DSoundChan() [%p]",this);
}
bool AttachHandler::received(Message &msg)
{
int more = 2;
String src(msg.getValue("source"));
if (src.null())
more--;
else if (!src.startSkip("dsound/",false))
src = "";
String cons(msg.getValue("consumer"));
if (cons.null())
more--;
else if (!cons.startSkip("dsound/",false))
cons = "";
if (src.null() && cons.null())
return false;
DataEndpoint *dd = static_cast<DataEndpoint*>(msg.userObject(YATOM("DataEndpoint")));
if (!dd) {
CallEndpoint *ch = static_cast<CallEndpoint*>(msg.userObject(YATOM("CallEndpoint")));
if (ch) {
dd = ch->setEndpoint();
if (!RefObject::alive(dd))
return false;
}
}
if (!dd) {
Debug(&__plugin,DebugWarn,"DSound attach request with no control or data channel!");
return false;
}
int rate = msg.getIntValue("rate",s_rate);
if (cons) {
DSoundConsumer* c = new DSoundConsumer(rate,msg.getBoolValue("stereo"));
dd->setConsumer(c);
c->deref();
Thread::msleep(50);
}
if (src) {
DSoundSource* s = new DSoundSource(rate);
dd->setSource(s);
s->deref();
Thread::msleep(50);
}
// Stop dispatching if we handled all requested
return !more;
}
bool SoundDriver::msgExecute(Message& msg, String& dest)
{
CallEndpoint* ch = static_cast<CallEndpoint*>(msg.userData());
if (ch) {
DSoundChan *ds = new DSoundChan(msg.getIntValue("rate",s_rate));
if (ch->connect(ds,msg.getValue("reason"))) {
msg.setParam("peerid",ds->id());
ds->deref();
}
else {
ds->destruct();
return false;
}
}
else {
Message m("call.route");
m.addParam("module",name());
String callto(msg.getValue("direct"));
if (callto.null()) {
const char *targ = msg.getValue("target");
if (!targ) {
Debug(&__plugin,DebugWarn,"DSound outgoing call with no target!");
return false;
}
callto = msg.getValue("caller");
if (callto.null())
callto << prefix() << dest;
m.addParam("called",targ);
m.addParam("caller",callto);
if (!Engine::dispatch(m)) {
Debug(&__plugin,DebugWarn,"DSound outgoing call but no route!");
return false;
}
callto = m.retValue();
m.retValue().clear();
}
m = "call.execute";
m.addParam("callto",callto);
DSoundChan *ds = new DSoundChan(msg.getIntValue("rate",8000));
m.setParam("targetid",ds->id());
m.userData(ds);
if (Engine::dispatch(m)) {
ds->deref();
return true;
}
Debug(&__plugin,DebugWarn,"DSound outgoing call not accepted!");
ds->destruct();
return false;
}
return true;
}
SoundDriver::SoundDriver()
: Driver("dsound","misc"),
m_handler(0)
{
Output("Loaded module DirectSound");
}
SoundDriver::~SoundDriver()
{
Output("Unloading module DirectSound");
channels().clear();
}
void SoundDriver::initialize()
{
Output("Initializing module DirectSound");
setup(0,true); // no need to install notifications
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) {
m_handler = new AttachHandler;
Engine::install(m_handler);
}
}
}; // anonymous namespace
#endif /* _WINDOWS */
/* vi: set ts=8 sw=4 sts=4 noet: */