commit a3841ecb3c665aa5e746a7056c215a4d6e102b2e
parent e75a6c086287c4deb14fac74078b5c9431807c7f
Author: fundamental <[email protected]>
Date: Wed, 9 Dec 2009 22:28:30 -0500
Nio: ALSA support enabled.
- Alsa is now functional, but there are several parameters, which are causing
unwanted latency in the output engine.
Diffstat:
5 files changed, 131 insertions(+), 360 deletions(-)
diff --git a/src/Nio/AlsaEngine.cpp b/src/Nio/AlsaEngine.cpp
@@ -33,12 +33,11 @@ AlsaEngine::AlsaEngine(OutMgr *out)
audio.handle = NULL;
audio.period_time = 0;
audio.samplerate = 0;
- audio.buffer_size = 0;
+ audio.buffer_size = SOUND_BUFFER_SIZE;//0;
audio.period_size = 0;
- audio.buffer_size = 0;
audio.alsaId = -1;
audio.pThread = 0;
-
+
// midi.handle = NULL;
// midi.alsaId = -1;
// midi.pThread = 0;
@@ -53,27 +52,8 @@ AlsaEngine::AlsaEngine(OutMgr *out)
bool AlsaEngine::openAudio()
{
- audio.device = "default";//config.cfg.audioDevice;
- audio.samplerate = config.cfg.SampleRate;
- audio.period_size = config.cfg.SoundBufferSize;
- audio.period_time = audio.period_size * 1000000.0f / audio.samplerate;
- alsaBad(snd_config_update_free_global(), "failed to update snd config");
- if (alsaBad(snd_pcm_open(&audio.handle, audio.device.c_str(),
- SND_PCM_STREAM_PLAYBACK, SND_PCM_NO_AUTO_CHANNELS),
- "failed to open alsa audio device:" + audio.device))
- goto bail_out;
- if (alsaBad(snd_pcm_nonblock(audio.handle, 0), "set blocking failed"))
- goto bail_out;
- if (!prepHwparams())
- goto bail_out;
- if (!prepSwparams())
- goto bail_out;
- //config.cfg.Samplerate = getSamplerate();
- //config.cfg.Buffersize = getBuffersize();
+ OpenStuff();
return true;
-bail_out:
- Close();
- return false;
}
@@ -112,24 +92,10 @@ bail_out:
void AlsaEngine::Close()
{
Stop();
- if (audio.handle != NULL)
- alsaBad(snd_pcm_close(audio.handle), "close pcm failed");
- audio.handle = NULL;
-// if (NULL != midi.handle)
-// if (snd_seq_close(midi.handle) < 0)
-// cerr << "Error closing Alsa midi connection" << endl;
-// midi.handle = NULL;
-}
-
-void AlsaEngine::out(const Stereo<Sample> smps)
-{
- pthread_mutex_lock(&outBuf_mutex);
- outBuf.push(smps);
- pthread_cond_signal(&outBuf_cv);
- pthread_mutex_unlock(&outBuf_mutex);
+ snd_pcm_drain(handle);
+ snd_pcm_close(handle);
}
-
string AlsaEngine::audioClientName()
{
string name = "zynaddsubfx";
@@ -146,125 +112,6 @@ string AlsaEngine::audioClientName()
// return name;
//}
-
-bool AlsaEngine::prepHwparams()
-{
- unsigned int buffer_time = audio.period_time * 4;
- unsigned int desired_samplerate = audio.samplerate;
- snd_pcm_format_t format = SND_PCM_FORMAT_S16; // Alsa appends _LE/_BE? hmmm
- snd_pcm_access_t axs = SND_PCM_ACCESS_MMAP_INTERLEAVED;
- snd_pcm_hw_params_t *hwparams = NULL;
- snd_pcm_hw_params_alloca(&hwparams);
- if (alsaBad(snd_pcm_hw_params_any(audio.handle, hwparams),
- "alsa audio no playback configurations available"))
- goto bail_out;
- if (alsaBad(snd_pcm_hw_params_set_periods_integer(audio.handle, hwparams),
- "alsa audio cannot restrict period size to integral value"))
- goto bail_out;
- if (!alsaBad(snd_pcm_hw_params_set_access(audio.handle, hwparams, axs),
- "alsa audio mmap not possible"))
- pcmWrite = &snd_pcm_mmap_writei;
- else
- {
- axs = SND_PCM_ACCESS_RW_INTERLEAVED;
- if (alsaBad(snd_pcm_hw_params_set_access(audio.handle, hwparams, axs),
- "alsa audio failed to set access, both mmap and rw failed"))
- goto bail_out;
- pcmWrite = &snd_pcm_writei;
- }
- if (alsaBad(snd_pcm_hw_params_set_format(audio.handle, hwparams, format),
- "alsa audio failed to set sample format"))
- goto bail_out;
- alsaBad(snd_pcm_hw_params_set_rate_resample(audio.handle, hwparams, 1),
- "alsa audio failed to set allow resample");
- if (alsaBad(snd_pcm_hw_params_set_rate_near(audio.handle, hwparams,
- &audio.samplerate, NULL),
- "alsa audio failed to set sample rate to "
- + stringFrom<int>(desired_samplerate)))
- goto bail_out;
- if (alsaBad(snd_pcm_hw_params_set_channels(audio.handle, hwparams, 2),
- "alsa audio failed to set channels to 2"))
- goto bail_out;
- if (!alsaBad(snd_pcm_hw_params_set_buffer_time_near(audio.handle, hwparams,
- &buffer_time, NULL), "initial buffer time setting failed"))
- {
- if (alsaBad(snd_pcm_hw_params_get_buffer_size(hwparams, &audio.buffer_size),
- "alsa audio failed to get buffer size"))
- goto bail_out;
- if (alsaBad(snd_pcm_hw_params_set_period_time_near(audio.handle, hwparams,
- &audio.period_time, NULL), "failed to set period time"))
- goto bail_out;
- if (alsaBad(snd_pcm_hw_params_get_period_size(hwparams, &audio.period_size,
- NULL), "alsa audio failed to get period size"))
- goto bail_out;
- }
- else
- {
- if (alsaBad(snd_pcm_hw_params_set_period_time_near(audio.handle, hwparams,
- &audio.period_time, NULL), "failed to set period time"))
- goto bail_out;
- audio.buffer_size = audio.period_size * 4;
- if (alsaBad(snd_pcm_hw_params_set_buffer_size_near(audio.handle, hwparams,
- &audio.buffer_size), "failed to set buffer size"))
- goto bail_out;
- }
- if (alsaBad(snd_pcm_hw_params(audio.handle, hwparams),
- "alsa audio failed to set hardware parameters"))
- goto bail_out;
- if (alsaBad(snd_pcm_hw_params_get_buffer_size(hwparams, &audio.buffer_size),
- "alsa audio failed to get buffer size"))
- goto bail_out;
- if (alsaBad(snd_pcm_hw_params_get_period_size(hwparams, &audio.period_size,
- NULL), "failed to get period size"))
- goto bail_out;
- return true;
-
-bail_out:
- if (audio.handle != NULL)
- Close();
- return false;
-}
-
-
-bool AlsaEngine::prepSwparams()
-{
- snd_pcm_sw_params_t *swparams;
- snd_pcm_sw_params_alloca(&swparams);
- snd_pcm_uframes_t boundary;
- if (alsaBad(snd_pcm_sw_params_current(audio.handle, swparams),
- "alsa audio failed to get swparams"))
- goto bail_out;
- if (alsaBad(snd_pcm_sw_params_get_boundary(swparams, &boundary),
- "alsa audio failed to get boundary"))
- goto bail_out;
- if (alsaBad(snd_pcm_sw_params_set_start_threshold(audio.handle, swparams,
- boundary + 1),
- "failed to set start threshold")) // explicit start, not auto start
- goto bail_out;
- if (alsaBad(snd_pcm_sw_params_set_stop_threshold(audio.handle, swparams,
- boundary),
- "alsa audio failed to set stop threshold"))
- goto bail_out;
- if (alsaBad(snd_pcm_sw_params(audio.handle, swparams),
- "alsa audio failed to set software parameters"))
- goto bail_out;
- return true;
-
-bail_out:
- return false;
-}
-
-
-bool AlsaEngine::alsaBad(int op_result, string err_msg)
-{
- bool isbad = (op_result < 0); // (op_result < 0) -> is bad -> return true
- if (isbad)
- cerr << "Error, alsa audio: " << err_msg << ": "
- << string(snd_strerror(op_result)) << endl;
- return isbad;
-}
-
-
void *AlsaEngine::_AudioThread(void *arg)
{
return (static_cast<AlsaEngine*>(arg))->AudioThread();
@@ -272,69 +119,20 @@ void *AlsaEngine::_AudioThread(void *arg)
void *AlsaEngine::AudioThread()
-{
- if (NULL == audio.handle)
- {
- cerr << "Error, null pcm handle into AlsaEngine::AudioThread" << endl;
- return NULL;
- }
- set_realtime();
- alsaBad(snd_pcm_start(audio.handle), "alsa audio pcm start failed");
- while (!threadStop)
- {
- cout << "AlsaEngine THREAD" << endl;
- const Stereo<Sample> smps = getNext();
-
- audio.pcm_state = snd_pcm_state(audio.handle);
- if (audio.pcm_state != SND_PCM_STATE_RUNNING)
- {
- switch (audio.pcm_state)
- {
- case SND_PCM_STATE_XRUN:
- case SND_PCM_STATE_SUSPENDED:
- if (!xrunRecover())
- break;
- // else fall through to ...
- case SND_PCM_STATE_SETUP:
- if (alsaBad(snd_pcm_prepare(audio.handle),
- "alsa audio pcm prepare failed"))
- break;
- case SND_PCM_STATE_PREPARED:
- alsaBad(snd_pcm_start(audio.handle), "pcm start failed");
- break;
- default:
- cout << "AlsaEngine::AudioThread, weird SND_PCM_STATE: "
- << audio.pcm_state << endl;
- break;
- }
- audio.pcm_state = snd_pcm_state(audio.handle);
- }
- if (audio.pcm_state == SND_PCM_STATE_RUNNING)
- {
- const short *tmp = interleave(smps);
- Write(tmp);
- delete [] tmp;
- }
- else
- {
- //config.cfg.verbose
- // && cerr << "Error, audio pcm still not RUNNING" << endl;
- cerr << "Error, audio pcm still not running";
- }
- }
+{
+ RunStuff();
return NULL;
}
-void AlsaEngine::Write(const short *InterleavedSmps)
+void AlsaEngine::Write(const short *InterleavedSmps,int size)
{
- snd_pcm_uframes_t towrite = getBuffersize();
+ snd_pcm_uframes_t towrite = size;//getBuffersize();
snd_pcm_sframes_t wrote = 0;
const short int *data = InterleavedSmps;
while (towrite > 0)
{
- //wrote = pcmWrite(audio.handle, &data, towrite);
- wrote = snd_pcm_writei(audio.handle, data, towrite);
+ wrote = pcmWrite(audio.handle, &data, towrite);
if (wrote >= 0)
{
if ((snd_pcm_uframes_t)wrote < towrite || wrote == -EAGAIN)
@@ -350,7 +148,7 @@ void AlsaEngine::Write(const short *InterleavedSmps)
switch (wrote)
{
case -EBADFD:
- alsaBad(-EBADFD, "alsa audio unfit for writing");
+ //alsaBad(-EBADFD, "alsa audio unfit for writing");
break;
case -EPIPE:
xrunRecover();
@@ -359,7 +157,7 @@ void AlsaEngine::Write(const short *InterleavedSmps)
Recover(wrote);
break;
default:
- alsaBad(wrote, "alsa audio, snd_pcm_writei ==> weird state");
+ //alsaBad(wrote, "alsa audio, snd_pcm_writei ==> weird state");
break;
}
wrote = 0;
@@ -379,14 +177,14 @@ bool AlsaEngine::Recover(int err)
isgood = true; // nuthin to see here
break;
case -ESTRPIPE:
- if (!alsaBad(snd_pcm_prepare(audio.handle),
- "Error, AlsaEngine failed to recover from suspend"))
- isgood = true;
+ // if (!alsaBad(snd_pcm_prepare(audio.handle),
+ // "Error, AlsaEngine failed to recover from suspend"))
+ // isgood = true;
break;
case -EPIPE:
- if (!alsaBad(snd_pcm_prepare(audio.handle),
- "Error, AlsaEngine failed to recover from underrun"))
- isgood = true;
+ // if (!alsaBad(snd_pcm_prepare(audio.handle),
+ // "Error, AlsaEngine failed to recover from underrun"))
+ // isgood = true;
break;
default:
break;
@@ -400,8 +198,8 @@ bool AlsaEngine::xrunRecover()
bool isgood = false;
if (audio.handle != NULL)
{
- if (!alsaBad(snd_pcm_drop(audio.handle), "pcm drop failed"))
- if (!alsaBad(snd_pcm_prepare(audio.handle), "pcm prepare failed"))
+ //if (!alsaBad(snd_pcm_drop(audio.handle), "pcm drop failed"))
+ // if (!alsaBad(snd_pcm_prepare(audio.handle), "pcm prepare failed"))
isgood = true;
;//config.cfg.verbose
// && cout << "Info, xrun recovery " << ((isgood) ? "good" : "not good")
@@ -416,12 +214,12 @@ bool AlsaEngine::Start(void)
int chk;
pthread_attr_t attr;
threadStop = false;
- if (NULL != audio.handle)
- {
+ //if (NULL != audio.handle)
+ //{
pthread_attr_init(&attr);
pthread_attr_setdetachstate(&attr, PTHREAD_CREATE_DETACHED);
pthread_create(&audio.pThread, &attr, _AudioThread, this);
- }
+ //}
// if (NULL != midi.handle)
// {
@@ -442,7 +240,7 @@ bail_out:
void AlsaEngine::Stop(void)
{
threadStop = true;
-
+
if (NULL != audio.handle && audio.pThread)
if (pthread_cancel(audio.pThread))
cerr << "Error, failed to cancel Alsa audio thread" << endl;
@@ -545,40 +343,11 @@ void AlsaEngine::Stop(void)
// return NULL;
//}
-const Stereo<Sample> AlsaEngine::getNext()
-{
- Stereo<Sample> ans;
- if(outBuf.empty())//fetch samples if possible
- {
- if(true)//FIXME care about locking state later on
- //mgr.requestSamples()!=-1)//samples are being prepared
- {
- manager->requestSamples();
- pthread_mutex_lock(&outBuf_mutex);
- pthread_cond_wait(&outBuf_cv, &outBuf_mutex);
- ans = outBuf.front();
- outBuf.pop();
- pthread_mutex_unlock(&outBuf_mutex);
- }
- }
- else
- {
- pthread_mutex_lock(&outBuf_mutex);
- ans = outBuf.front();
- outBuf.pop();
- pthread_mutex_unlock(&outBuf_mutex);
- }
- return ans;
-}
-
-
const short *AlsaEngine::interleave(const Stereo<Sample> smps)const
{
- //hm, this seems less than optimum
- //TODO remove this excessive allocation/deallocation once things are
- //integrated
- short *shortInterleaved = new short[smps.l().size()*2+1002];//over allocation
- /**\todo TODO fix overallocation*/
+ /**\todo TODO fix repeated allocation*/
+ short *shortInterleaved = new short[smps.l().size()*2];//over allocation
+ memset(shortInterleaved,0,smps.l().size()*2*sizeof(short));
int idx = 0;//possible off by one error here
double scaled;
for (int frame = 0; frame < smps.l().size(); ++frame)
@@ -588,4 +357,85 @@ const short *AlsaEngine::interleave(const Stereo<Sample> smps)const
scaled = smps.r()[frame] * (8.0 * 0x10000000);
shortInterleaved[idx++] = (short int)(lrint(scaled) >> 16);
}
+ return shortInterleaved;
+}
+
+
+void AlsaEngine::OpenStuff()
+{
+ /* Open PCM device for playback. */
+ handle=NULL;
+ rc = snd_pcm_open(&handle, "default",
+ SND_PCM_STREAM_PLAYBACK, 0);
+ if (rc < 0) {
+ fprintf(stderr,
+ "unable to open pcm device: %s\n",
+ snd_strerror(rc));
+ exit(1);
+ }
+
+ /* Allocate a hardware parameters object. */
+ snd_pcm_hw_params_alloca(¶ms);
+
+ /* Fill it in with default values. */
+ snd_pcm_hw_params_any(handle, params);
+
+ /* Set the desired hardware parameters. */
+
+ /* Interleaved mode */
+ snd_pcm_hw_params_set_access(handle, params,
+ SND_PCM_ACCESS_RW_INTERLEAVED);
+
+ /* Signed 16-bit little-endian format */
+ snd_pcm_hw_params_set_format(handle, params,
+ SND_PCM_FORMAT_S16_LE);
+
+ /* Two channels (stereo) */
+ snd_pcm_hw_params_set_channels(handle, params, 2);
+
+ /* 44100 bits/second sampling rate (CD quality)
+ * \TODO make this dynamic*/
+ val = 44100;
+ snd_pcm_hw_params_set_rate_near(handle, params,
+ &val, NULL);//&dir);
+
+ frames = 32;
+ snd_pcm_hw_params_set_period_size_near(handle,
+ params, &frames, NULL);//&dir);
+
+ /* Write the parameters to the driver */
+ rc = snd_pcm_hw_params(handle, params);
+ if (rc < 0) {
+ fprintf(stderr,
+ "unable to set hw parameters: %s\n",
+ snd_strerror(rc));
+ exit(1);
+ }
+
+ /* Set buffer size (in frames). The resulting latency is given by */
+ /* latency = periodsize * periods / (rate * bytes_per_frame) */
+ snd_pcm_hw_params_set_buffer_size(handle, params, SOUND_BUFFER_SIZE);
+
+ /* Use a buffer large enough to hold one period */
+ snd_pcm_hw_params_get_period_size(params, &frames, &dir);
+
+ snd_pcm_hw_params_get_period_time(params, &val, &dir);
+}
+
+void AlsaEngine::RunStuff()
+{
+ while (!threadStop()) {
+ buffer = interleave(getNext());
+ rc = snd_pcm_writei(handle, buffer, SOUND_BUFFER_SIZE);
+ delete buffer;
+ if (rc == -EPIPE) {
+ /* EPIPE means underrun */
+ cerr << "underrun occurred" << endl;
+ snd_pcm_prepare(handle);
+ }
+ else if (rc < 0)
+ cerr << "error from writei: " << snd_strerror(rc) << endl;
+ else if (rc != (int)frames)
+ cerr << "short write, write " << rc << "frames" << endl;
+ }
}
diff --git a/src/Nio/AlsaEngine.h b/src/Nio/AlsaEngine.h
@@ -42,7 +42,6 @@ class AlsaEngine : public AudioOut//, MidiIn
bool Start();
void Stop();
void Close();
- void out(const Stereo<Sample> smps);
unsigned int getSamplerate() { return audio.samplerate; };
unsigned int getBuffersize() { return audio.period_size; };
@@ -61,7 +60,7 @@ class AlsaEngine : public AudioOut//, MidiIn
private:
bool prepHwparams();
bool prepSwparams();
- void Write(const short *InterleavedSmps);
+ void Write(const short *InterleavedSmps, int size);
bool Recover(int err);
bool xrunRecover();
bool alsaBad(int op_result, std::string err_msg);
@@ -72,7 +71,7 @@ class AlsaEngine : public AudioOut//, MidiIn
snd_pcm_uframes_t nframes);
/**Get the next sample for output.*/
- const Stereo<Sample> getNext();
+ //const Stereo<Sample> getNext();
/**Interleave Samples. \todo move this to util*/
const short *interleave(const Stereo<Sample> smps) const;
@@ -95,7 +94,21 @@ class AlsaEngine : public AudioOut//, MidiIn
// pthread_t pThread;
//} midi;
- bool threadStop;
+ //from alsa example
+ long loops;
+ int rc;
+ int size;
+ snd_pcm_t *handle;
+ snd_pcm_hw_params_t *params;
+ unsigned int val;
+ int dir;
+ snd_pcm_uframes_t frames;
+ const short *buffer;
+
+ void RunStuff();
+ void OpenStuff();
+
+ //bool threadStop;
//outside audio interface
queue<Stereo<Sample> > outBuf;
diff --git a/src/Nio/AudioOut.cpp b/src/Nio/AudioOut.cpp
@@ -33,91 +33,6 @@ AudioOut::AudioOut(OutMgr *out)
pthread_cond_init (&outBuf_cv, NULL);
}
-//bool AudioOut::prepAudiobuffers(unsigned int nframes, bool with_interleaved)
-//{
-// if (nframes > 0)
-// {
-// buffersize = nframes;
-// outl = new float[buffersize];
-// outr = new float[buffersize];
-// if (outl == NULL || outr == NULL)
-// goto bail_out;
-// memset(outl, 0, buffersize * sizeof(float));
-// memset(outr, 0, buffersize * sizeof(float));
-// if (with_interleaved)
-// {
-// shortInterleaved = new short int[buffersize * 2];
-// if (shortInterleaved == NULL)
-// goto bail_out;
-// memset(shortInterleaved, 0, buffersize * 2 * sizeof(short int));
-// }
-// return true;
-// }
-//
-//bail_out:
-// cerr << "Error, failed to allocate audio buffers, size " << buffersize << endl;
-// if (outl != NULL)
-// delete [] outl;
-// outl = NULL;
-// if (outr != NULL)
-// delete [] outr;
-// outr = NULL;
-// if (with_interleaved)
-// {
-// if (shortInterleaved != NULL)
-// delete [] shortInterleaved;
-// shortInterleaved = NULL;
-// }
-// return false;
-//}
-//
-//
-//bool AudioOut::getAudio(bool lockrequired)
-//{
-// if (NULL != master && outr != NULL && outl != NULL)
-// return master->MasterAudio(outl, outr, lockrequired);
-// return false;
-//}
-//
-//
-//bool AudioOut::getAudioInterleaved(bool lockrequired)
-//{
-// if (shortInterleaved != NULL)
-// {
-// if (getAudio(lockrequired))
-// {
-// int idx = 0;
-// double scaled;
-// for (int frame = 0; frame < buffersize; ++frame)
-// { // with a nod to libsamplerate ...
-// scaled = outl[frame] * (8.0 * 0x10000000);
-// shortInterleaved[idx++] = (short int)(lrint(scaled) >> 16);
-// scaled = outr[frame] * (8.0 * 0x10000000);
-// shortInterleaved[idx++] = (short int)(lrint(scaled) >> 16);
-// }
-// return true;
-// }
-// }
-// return false;
-//}
-//
-//
-//void AudioOut::silenceBuffers()
-//{
-// memset(outl, 0, buffersize * sizeof(float));
-// memset(outr, 0, buffersize * sizeof(float));
-//}
-//
-//
-//void AudioOut::dimBuffers()
-//{
-// for (int i = 0; i < buffersize; ++i)
-// {
-// outl[i] *= 0.84033613; // -1.5dB
-// outr[i] *= 0.84033613;
-// }
-//}
-
void AudioOut::out(const Stereo<Sample> smps)
{
pthread_mutex_lock(&outBuf_mutex);
@@ -134,7 +49,6 @@ const Stereo<Sample> AudioOut::getNext()
pthread_mutex_unlock(&outBuf_mutex);
if(isEmpty)//fetch samples if possible
{
- //cerr << manager << endl;
if(manager->getRunning()<4)
manager->requestSamples();
if(true)
diff --git a/src/Nio/OutMgr.cpp b/src/Nio/OutMgr.cpp
@@ -23,7 +23,6 @@ OutMgr::OutMgr(Master *nmaster)
//initialize mutex
pthread_mutex_init(&mutex, NULL);
pthread_mutex_init(&processing, NULL);
- //pthread_mutex_init(&request_m, NULL);
pthread_cond_init(&needsProcess, NULL);
//init samples
outr = new REALTYPE[SOUND_BUFFER_SIZE];
@@ -142,18 +141,12 @@ void OutMgr::remove(AudioOut *out)
int OutMgr::getRunning()
{
- //int tmp;
- //pthread_mutex_lock(&request_m);
- //tmp=eumRequests;
- //pthread_mutex_unlock(&request_m);
- return numRequests();//tmp;
+ return numRequests();
}
int OutMgr::requestSamples()
{
- //pthread_mutex_lock(&request_m);
++numRequests;
- //pthread_mutex_unlock(&request_m);
pthread_mutex_lock(&processing);
pthread_cond_signal(&needsProcess);
diff --git a/src/main.cpp b/src/main.cpp
@@ -687,11 +687,12 @@ int main(int argc, char *argv[])
sysOut=NULL;
sysOut = new OutMgr(master);
if(sysOut);
- //AlsaEngine *tmp = new AlsaEngine();
- AudioOut *tmp = new OssEngine(sysOut);
- if(tmp);
+ AudioOut *tmp = new AlsaEngine(sysOut);
+ tmp->openAudio();
//tmp->openAudio();
- sysOut->add(tmp);
+ //AudioOut *tmp = new OssEngine(sysOut);
+ if(tmp)
+ sysOut->add(tmp);
sysOut->run();
#endif
