// Modified by:
// Created: 25/10/98
// RCS-ID: $Id$
-// Copyright: (c) Julian Smart, Vaclav Slavik
+// Copyright: (c) Julian Smart, Open Source Applications Foundation
// Licence: wxWindows licence
/////////////////////////////////////////////////////////////////////////////
#pragma hdrstop
#endif
-#if wxUSE_WAVE
+#if wxUSE_SOUND
#include <stdio.h>
#include <unistd.h>
private:
int OpenDSP(const wxSoundData *data);
- bool InitDSP(int dev, int iDataBits, int iChannel,
- unsigned long ulSamplingRate);
+ bool InitDSP(int dev, const wxSoundData *data);
int m_DSPblkSize; // Size of the DSP buffer
+ bool m_needConversion;
};
bool wxSoundBackendOSS::IsAvailable() const
return false;
ioctl(dev, SNDCTL_DSP_SYNC, 0);
-
+
do
{
bool play = true;
}
i= (int)((l + m_DSPblkSize) < datasize ?
- m_DSPblkSize : (datasize - l));
+ m_DSPblkSize : (datasize - l));
if (write(dev, &data->m_data[l], i) != i)
{
play = false;
l += i;
} while (play && l < datasize);
} while (flags & wxSOUND_LOOP);
-
- close(dev);
+ close(dev);
return true;
}
if ((dev = open(AUDIODEV, O_WRONLY, 0)) <0)
return -1;
- if (!InitDSP(dev,
- (int)data->m_bitsPerSample,
- data->m_channels == 1 ? 0 : 1,
- data->m_samplingRate))
+ if (!InitDSP(dev, data) || m_needConversion)
{
close(dev);
return -1;
return dev;
}
-bool wxSoundBackendOSS::InitDSP(int dev, int iDataBits, int iChannel,
- unsigned long ulSamplingRate)
+
+bool wxSoundBackendOSS::InitDSP(int dev, const wxSoundData *data)
{
- if (ioctl(dev, SNDCTL_DSP_GETBLKSIZE, &m_DSPblkSize) < 0)
- return false;
- wxLogTrace(_T("sound"), _T("OSS block size: %i"), m_DSPblkSize);
- if (m_DSPblkSize < 4096 || m_DSPblkSize > 65536)
+ unsigned tmp;
+
+ // Reset the dsp
+ if (ioctl(dev, SNDCTL_DSP_RESET, 0) < 0)
+ {
+ wxLogTrace(_T("sound"), _T("unable to reset dsp"));
return false;
- if (ioctl(dev, SNDCTL_DSP_SAMPLESIZE, &iDataBits) < 0)
+ }
+
+ m_needConversion = false;
+
+ tmp = data->m_bitsPerSample;
+ if (ioctl(dev, SNDCTL_DSP_SAMPLESIZE, &tmp) < 0)
+ {
+ wxLogTrace(_T("sound"), _T("IOCTL failure (SNDCTL_DSP_SAMPLESIZE)"));
return false;
- if (ioctl(dev, SNDCTL_DSP_STEREO, &iChannel) < 0)
+ }
+ if (tmp != data->m_bitsPerSample)
+ {
+ wxLogTrace(_T("sound"),
+ _T("Unable to set DSP sample size to %d (wants %d)"),
+ data->m_bitsPerSample, tmp);
+ m_needConversion = true;
+ }
+
+ unsigned stereo = data->m_channels == 1 ? 0 : 1;
+ tmp = stereo;
+ if (ioctl(dev, SNDCTL_DSP_STEREO, &tmp) < 0)
+ {
+ wxLogTrace(_T("sound"), _T("IOCTL failure (SNDCTL_DSP_STEREO)"));
return false;
- if (ioctl(dev, SNDCTL_DSP_SPEED, &ulSamplingRate) < 0)
+ }
+ if (tmp != stereo)
+ {
+ wxLogTrace(_T("sound"), _T("Unable to set DSP to %s."), stereo? _T("stereo"):_T("mono"));
+ m_needConversion = true;
+ }
+
+ tmp = data->m_samplingRate;
+ if (ioctl(dev, SNDCTL_DSP_SPEED, &tmp) < 0)
+ {
+ wxLogTrace(_T("sound"), _T("IOCTL failure (SNDCTL_DSP_SPEED)"));
+ return false;
+ }
+ if (tmp != data->m_samplingRate)
+ {
+ // If the rate the sound card is using is not within 1% of what the
+ // data specified then override the data setting. The only reason not
+ // to always override this is because of clock-rounding
+ // problems. Sound cards will sometimes use things like 44101 when you
+ // ask for 44100. No need overriding this and having strange output
+ // file rates for something that we can't hear anyways.
+ if (data->m_samplingRate - tmp > (tmp * .01) ||
+ tmp - data->m_samplingRate > (tmp * .01)) {
+ wxLogTrace(_T("sound"),
+ _T("Unable to set DSP sampling rate to %d (wants %d)"),
+ data->m_samplingRate, tmp);
+ m_needConversion = true;
+ }
+ }
+
+ // Do this last because some drivers can adjust the buffer sized based on
+ // the sampling rate, etc.
+ if (ioctl(dev, SNDCTL_DSP_GETBLKSIZE, &m_DSPblkSize) < 0)
+ {
+ wxLogTrace(_T("sound"), _T("IOCTL failure (SNDCTL_DSP_GETBLKSIZE)"));
return false;
+ }
return true;
}
-
+
#endif // HAVE_SYS_SOUNDCARD_H
// ----------------------------------------------------------------------------
void wxSoundSyncOnlyAdaptor::Stop()
{
wxLogTrace(_T("sound"), _T("asking audio to stop"));
+
+#if wxUSE_THREADS
// tell the player thread (if running) to stop playback ASAP:
m_status.m_stopRequested = true;
m_mutexRightToPlay.Lock();
m_mutexRightToPlay.Unlock();
wxLogTrace(_T("sound"), _T("audio was stopped"));
+#endif
}
bool wxSoundSyncOnlyAdaptor::IsPlaying() const
{
+#if wxUSE_THREADS
return m_status.m_playing;
+#else
+ return FALSE;
+#endif
}
{
// FIXME -- make this fully dynamic when plugins architecture is in
// place
-#ifdef HAVE_SYS_SOUNDCARD_H
- ms_backend = new wxSoundBackendOSS();
- if (!ms_backend->IsAvailable())
- {
- wxDELETE(ms_backend);
- }
-#endif
-
#if wxUSE_LIBSDL
- if (!ms_backend)
+ //if (!ms_backend)
{
#if !wxUSE_PLUGINS
ms_backend = wxCreateSoundBackendSDL();
}
#endif
+#ifdef HAVE_SYS_SOUNDCARD_H
+ if (!ms_backend)
+ {
+ ms_backend = new wxSoundBackendOSS();
+ if (!ms_backend->IsAvailable())
+ {
+ wxDELETE(ms_backend);
+ }
+ }
+#endif
+
if (!ms_backend)
ms_backend = new wxSoundBackendNull();
}
}
-bool wxSound::DoPlay(unsigned flags)
+bool wxSound::DoPlay(unsigned flags) const
{
- wxASSERT_MSG( (flags & wxSOUND_LOOP) == 0 || (flags & wxSOUND_ASYNC) != 0,
- _T("sound can only be looped asynchronously") );
wxCHECK_MSG( IsOk(), false, _T("Attempt to play invalid wave data") );
EnsureBackend();
projects / wxWidgets.git / blobdiff