void I2_SetPitch(sndsource_t *src, float pitch)
{
int newfreq = src->freq * pitch;
if(newfreq < DSBFREQUENCY_MIN) newfreq = DSBFREQUENCY_MIN;
if(newfreq > DSBFREQUENCY_MAX) newfreq = DSBFREQUENCY_MAX;
IDirectSoundBuffer_SetFrequency(src->source, newfreq);
}
void I_UpdateSoundParams(int channel, int vol, int sep, int pitch)
{
// proff 07/04/98: Added for CYGWIN32 compatibility
#ifdef HAVE_LIBDSOUND
int DSB_Status;
if (noDSound == true)
return;
// proff 07/26/98: Added volume check
if (vol==0)
{
IDirectSoundBuffer_Stop(lpSecondaryDSB[channel]);
return;
}
IDirectSoundBuffer_SetVolume(lpSecondaryDSB[channel],VOL(vol));
IDirectSoundBuffer_SetPan(lpSecondaryDSB[channel],SEP(sep));
IDirectSoundBuffer_SetFrequency
(lpSecondaryDSB[channel], ChannelInfo[channel].samplerate+PITCH(pitch));
if (ChannelInfo[channel].playing == true)
{
IDirectSoundBuffer_GetStatus(lpSecondaryDSB[channel], &DSB_Status);
if ((DSB_Status & DSBSTATUS_PLAYING) == 0)
IDirectSoundBuffer_Play(lpSecondaryDSB[channel], 0, 0, 0);
}
#endif // HAVE_LIBDSOUND
}
static HRESULT WINAPI DSoundRender_PrepareReceive(BaseRenderer *iface, IMediaSample *pSample)
{
DSoundRenderImpl *This = impl_from_BaseRenderer(iface);
HRESULT hr;
AM_MEDIA_TYPE *amt;
if (IMediaSample_GetMediaType(pSample, &amt) == S_OK)
{
AM_MEDIA_TYPE *orig = &This->renderer.pInputPin->pin.mtCurrent;
WAVEFORMATEX *origfmt = (WAVEFORMATEX *)orig->pbFormat;
WAVEFORMATEX *newfmt = (WAVEFORMATEX *)amt->pbFormat;
if (origfmt->wFormatTag == newfmt->wFormatTag &&
origfmt->nChannels == newfmt->nChannels &&
origfmt->nBlockAlign == newfmt->nBlockAlign &&
origfmt->wBitsPerSample == newfmt->wBitsPerSample &&
origfmt->cbSize == newfmt->cbSize)
{
if (origfmt->nSamplesPerSec != newfmt->nSamplesPerSec)
{
hr = IDirectSoundBuffer_SetFrequency(This->dsbuffer,
newfmt->nSamplesPerSec);
if (FAILED(hr))
return VFW_E_TYPE_NOT_ACCEPTED;
FreeMediaType(orig);
CopyMediaType(orig, amt);
IMediaSample_SetMediaType(pSample, NULL);
}
}
else
return VFW_E_TYPE_NOT_ACCEPTED;
}
return S_OK;
}
void windows_osd_interface::update_audio_stream(const INT16 *buffer, int samples_this_frame)
{
int bytes_this_frame = samples_this_frame * stream_format.nBlockAlign;
DWORD play_position, write_position;
HRESULT result;
// if no sound, there is no buffer
if (stream_buffer == NULL)
return;
#ifdef USE_AUDIO_SYNC
// if we are active, update the sampling frequency
if (audio_sync && machine().video().speed_percent() > 0.0f)
{
IDirectSoundBuffer_SetFrequency(stream_buffer, machine().sample_rate() * machine().video().speed_percent());
}
#endif /* USE_AUDIO_SYNC */
// determine the current play position
result = IDirectSoundBuffer_GetCurrentPosition(stream_buffer, &play_position, &write_position);
if (result == DS_OK)
{
DWORD stream_in;
//DWORD orig_write = write_position;
// normalize the write position so it is always after the play position
if (write_position < play_position)
write_position += stream_buffer_size;
// normalize the stream in position so it is always after the write position
stream_in = stream_buffer_in;
if (stream_in < write_position)
stream_in += stream_buffer_size;
// now we should have, in order:
// <------pp---wp---si--------------->
// if we're between play and write positions, then bump forward, but only in full chunks
while (stream_in < write_position)
{
//printf("Underflow: PP=%d WP=%d(%d) SI=%d(%d) BTF=%d\n", (int)play_position, (int)write_position, (int)orig_write, (int)stream_in, (int)stream_buffer_in, (int)bytes_this_frame);
buffer_underflows++;
stream_in += bytes_this_frame;
}
// if we're going to overlap the play position, just skip this chunk
if (stream_in + bytes_this_frame > play_position + stream_buffer_size)
{
//printf("Overflow: PP=%d WP=%d(%d) SI=%d(%d) BTF=%d\n", (int)play_position, (int)write_position, (int)orig_write, (int)stream_in, (int)stream_buffer_in, (int)bytes_this_frame);
buffer_overflows++;
return;
}
// now we know where to copy; let's do it
stream_buffer_in = stream_in % stream_buffer_size;
copy_sample_data(buffer, bytes_this_frame);
}
}
static BOOL ModifyChannel( Channel *channel, geSound_Cfg *cfg )
{
int Error, Vol, Pan, Freq;
assert(channel);
if( !cfg )
return( TRUE );
ClearDupBuffers(channel);
if( cfg->Volume != channel->cfg.Volume )
{
Vol = (DWORD)((1.0 - cfg->Volume ) * DSBVOLUME_MIN);
Error = IDirectSoundBuffer_SetVolume(channel->buffer, Vol);
if (Error != DS_OK)
{
geErrorLog_Add(GE_ERR_DS_ERROR, NULL);
return FALSE;
}
channel->cfg.Volume = cfg->Volume;
}
if( cfg->Pan != channel->cfg.Pan )
{
Pan = (int)(cfg->Pan * DSBPAN_RIGHT);
Error = IDirectSoundBuffer_SetPan(channel->buffer, Pan);
if (Error != DS_OK)
{
geErrorLog_Add(GE_ERR_DS_ERROR, NULL);
return FALSE;
}
channel->cfg.Pan = cfg->Pan;
}
if( cfg->Frequency != channel->cfg.Frequency )
{
Freq = (DWORD)(channel->BaseFreq * cfg->Frequency);
if(Freq < 0)
Freq = 0;
Error = IDirectSoundBuffer_SetFrequency(channel->buffer, Freq);
if (Error != DS_OK)
{
geErrorLog_Add(GE_ERR_DS_ERROR, NULL);
return FALSE;
}
channel->cfg.Frequency = cfg->Frequency;
}
return TRUE;
}
static HRESULT WINAPI DSoundRender_Receive(BaseInputPin *pin, IMediaSample * pSample)
{
DSoundRenderImpl *This = (DSoundRenderImpl*)pin->pin.pinInfo.pFilter;
LPBYTE pbSrcStream = NULL;
LONG cbSrcStream = 0;
REFERENCE_TIME tStart, tStop;
HRESULT hr;
AM_MEDIA_TYPE *amt;
TRACE("%p %p\n", pin, pSample);
/* Slightly incorrect, Pause completes when a frame is received so we should signal
* pause completion here, but for sound playing a single frame doesn't make sense
*/
EnterCriticalSection(&This->filter.csFilter);
if (This->pInputPin->end_of_stream || This->pInputPin->flushing)
{
LeaveCriticalSection(&This->filter.csFilter);
return S_FALSE;
}
if (This->filter.state == State_Stopped)
{
LeaveCriticalSection(&This->filter.csFilter);
return VFW_E_WRONG_STATE;
}
if (IMediaSample_GetMediaType(pSample, &amt) == S_OK)
{
AM_MEDIA_TYPE *orig = &This->pInputPin->pin.mtCurrent;
WAVEFORMATEX *origfmt = (WAVEFORMATEX *)orig->pbFormat;
WAVEFORMATEX *newfmt = (WAVEFORMATEX *)amt->pbFormat;
if (origfmt->wFormatTag == newfmt->wFormatTag &&
origfmt->nChannels == newfmt->nChannels &&
origfmt->nBlockAlign == newfmt->nBlockAlign &&
origfmt->wBitsPerSample == newfmt->wBitsPerSample &&
origfmt->cbSize == newfmt->cbSize)
{
if (origfmt->nSamplesPerSec != newfmt->nSamplesPerSec)
{
hr = IDirectSoundBuffer_SetFrequency(This->dsbuffer,
newfmt->nSamplesPerSec);
if (FAILED(hr))
{
LeaveCriticalSection(&This->filter.csFilter);
return VFW_E_TYPE_NOT_ACCEPTED;
}
FreeMediaType(orig);
CopyMediaType(orig, amt);
IMediaSample_SetMediaType(pSample, NULL);
}
}
else
{
LeaveCriticalSection(&This->filter.csFilter);
return VFW_E_TYPE_NOT_ACCEPTED;
}
}
hr = IMediaSample_GetPointer(pSample, &pbSrcStream);
if (FAILED(hr))
{
ERR("Cannot get pointer to sample data (%x)\n", hr);
LeaveCriticalSection(&This->filter.csFilter);
return hr;
}
hr = IMediaSample_GetTime(pSample, &tStart, &tStop);
if (FAILED(hr))
ERR("Cannot get sample time (%x)\n", hr);
else
MediaSeekingPassThru_RegisterMediaTime(This->seekthru_unk, tStart);
if (This->rtLastStop != tStart && (IMediaSample_IsDiscontinuity(pSample) == S_FALSE))
WARN("Unexpected discontinuity: Last: %u.%03u, tStart: %u.%03u\n",
(DWORD)(This->rtLastStop / 10000000), (DWORD)((This->rtLastStop / 10000)%1000),
(DWORD)(tStart / 10000000), (DWORD)((tStart / 10000)%1000));
This->rtLastStop = tStop;
if (IMediaSample_IsPreroll(pSample) == S_OK)
{
TRACE("Preroll!\n");
LeaveCriticalSection(&This->filter.csFilter);
return S_OK;
}
if (This->filter.state == State_Paused)
{
SetEvent(This->state_change);
LeaveCriticalSection(&This->filter.csFilter);
WaitForSingleObject(This->blocked, INFINITE);
EnterCriticalSection(&This->filter.csFilter);
if (This->filter.state == State_Stopped)
{
LeaveCriticalSection(&This->filter.csFilter);
return VFW_E_WRONG_STATE;
}
if (This->filter.state == State_Paused)
{
/* Assuming we return because of flushing */
TRACE("Flushing\n");
LeaveCriticalSection(&This->filter.csFilter);
return S_OK;
}
SetEvent(This->state_change);
}
cbSrcStream = IMediaSample_GetActualDataLength(pSample);
TRACE("Sample data ptr = %p, size = %d\n", pbSrcStream, cbSrcStream);
#if 0 /* For debugging purpose */
{
int i;
for(i = 0; i < cbSrcStream; i++)
{
if ((i!=0) && !(i%16))
TRACE("\n");
TRACE("%02x ", pbSrcStream[i]);
}
TRACE("\n");
}
#endif
hr = DSoundRender_SendSampleData(This, pbSrcStream, cbSrcStream);
SetEvent(This->state_change);
LeaveCriticalSection(&This->filter.csFilter);
return hr;
}
int I_StartSound(sfxinfo_t *sound, int vol, int sep, int pitch, int pri)
{
int channel=0;
// proff 07/04/98: Added for CYGWIN32 compatibility
#ifdef HAVE_LIBDSOUND
HRESULT error;
char *snddata;
int sndlength;
if (noDSound == true)
return channel;
// load sound data if we have not already
I_CacheSound(sound);
// find a free channel
channel = I_GetFreeChannel();
// proff 07/26/98: Added volume check
// proff 10/31/98: Added Stop before updating sound-data
error = IDirectSoundBuffer_Stop(lpSecondaryDSB[channel]);
ChannelInfo[channel].playing = false;
if (vol==0)
return channel;
snddata = sound->data;
ChannelInfo[channel].samplerate = (snddata[3] << 8) + snddata[2];
// proff 10/31/98: Use accurate time for this one
ChannelInfo[channel].endtime =
I_GetTime_RealTime() +
(sound->length * 35) / ChannelInfo[channel].samplerate + 1;
// skip past header
snddata += 8;
sndlength = sound->length - 8;
error = IDirectSoundBuffer_SetCurrentPosition(lpSecondaryDSB[channel],0);
// proff 11/09/98: Added for a slight speedup
if (sound != ChannelInfo[channel].sfx)
{
DWORD *hand1,*hand2;
DWORD len1,len2;
ChannelInfo[channel].sfx = sound;
error = IDirectSoundBuffer_Lock(lpSecondaryDSB[channel],0,65535,
&hand1,&len1,&hand2,&len2,
DSBLOCK_FROMWRITECURSOR);
if (len1 >= sndlength)
{
memset(hand1, 128, len1);
memcpy(hand1, snddata , sndlength);
memset(hand2, 128, len2);
}
else
{
memcpy(hand1, snddata, len1);
memcpy(hand2, &((char *)snddata)[len1], sndlength-len1);
}
error = IDirectSoundBuffer_Unlock
(lpSecondaryDSB[channel], hand1, len1, hand2, len2);
}
IDirectSoundBuffer_SetVolume(lpSecondaryDSB[channel], VOL(vol));
IDirectSoundBuffer_SetPan(lpSecondaryDSB[channel], SEP(sep));
IDirectSoundBuffer_SetFrequency(lpSecondaryDSB[channel],
ChannelInfo[channel].samplerate+PITCH(pitch));
error = IDirectSoundBuffer_Play(lpSecondaryDSB[channel], 0, 0, 0);
ChannelInfo[channel].playing = true;
#endif // HAVE_LIBDSOUND
return channel;
}
void SOUND_set_frequency(int sound, int freq)
{
IDirectSoundBuffer_SetFrequency(sample_buffers[sound],freq) ;
}
