public IALBuffer ConvertStereoToMono(IALBuffer buffer)
{
OpenALBuffer buf = buffer as OpenALBuffer;
int bufLen, bits;
AL10.alGetBufferi(
buf.Handle,
AL10.AL_SIZE,
out bufLen
);
AL10.alGetBufferi(
buf.Handle,
AL10.AL_BITS,
out bits
);
bits /= 8;
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
byte[] data = new byte[bufLen];
GCHandle dataHandle = GCHandle.Alloc(data, GCHandleType.Pinned);
IntPtr dataPtr = dataHandle.AddrOfPinnedObject();
ALEXT.alGetBufferSamplesSOFT(
buf.Handle,
0,
bufLen / bits / 2,
ALEXT.AL_STEREO_SOFT,
bits == 2 ? ALEXT.AL_SHORT_SOFT : ALEXT.AL_BYTE_SOFT,
dataPtr
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
byte[] monoData = new byte[bufLen / 2];
GCHandle monoHandle = GCHandle.Alloc(monoData, GCHandleType.Pinned);
IntPtr monoPtr = monoHandle.AddrOfPinnedObject();
unsafe
{
if (bits == 2)
{
short *src = (short *)dataPtr;
short *dst = (short *)monoPtr;
for (int i = 0; i < monoData.Length / 2; i += 1)
{
dst[i] = (short)(((int)src[0] + (int)src[1]) / 2);
src += 2;
}
}
else
{
sbyte *src = (sbyte *)dataPtr;
sbyte *dst = (sbyte *)monoPtr;
for (int i = 0; i < monoData.Length; i += 1)
{
dst[i] = (sbyte)(((short)src[0] + (short)src[1]) / 2);
src += 2;
}
}
}
monoHandle.Free();
dataHandle.Free();
data = null;
return(GenBuffer(
monoData,
(uint)buf.SampleRate,
1,
0,
0,
false,
(uint)bits - 1
));
}
public void GetBufferData(
IALSource source,
IALBuffer[] buffer,
IntPtr samples,
int samplesLen,
AudioChannels channels
)
{
int copySize1 = samplesLen / (int)channels;
int copySize2 = 0;
// Where are we now?
int offset;
AL10.alGetSourcei(
(source as OpenALSource).Handle,
AL11.AL_SAMPLE_OFFSET,
out offset
);
// Is that longer than what the active buffer has left...?
uint buf = (buffer[0] as OpenALBuffer).Handle;
int len;
AL10.alGetBufferi(
buf,
AL10.AL_SIZE,
out len
);
len /= 2; // FIXME: Assuming 16-bit!
len /= (int)channels;
if (offset > len)
{
copySize2 = copySize1;
copySize1 = 0;
offset -= len;
}
else if (offset + copySize1 > len)
{
copySize2 = copySize1 - (len - offset);
copySize1 = (len - offset);
}
// Copy!
if (copySize1 > 0)
{
ALEXT.alGetBufferSamplesSOFT(
buf,
offset,
copySize1,
channels == AudioChannels.Stereo ?
ALEXT.AL_STEREO_SOFT :
ALEXT.AL_MONO_SOFT,
ALEXT.AL_FLOAT_SOFT,
samples
);
offset = 0;
}
if (buffer.Length > 1 && copySize2 > 0)
{
ALEXT.alGetBufferSamplesSOFT(
(buffer[1] as OpenALBuffer).Handle,
0,
copySize2,
channels == AudioChannels.Stereo ?
ALEXT.AL_STEREO_SOFT :
ALEXT.AL_MONO_SOFT,
ALEXT.AL_FLOAT_SOFT,
samples + (copySize1 * (int)channels)
);
}
}
public unsafe IALBuffer ConvertStereoToMono(IALBuffer buffer)
{
OpenALBuffer buf = buffer as OpenALBuffer;
int bufLen, bits;
AL10.alGetBufferi(
buf.Handle,
AL10.AL_SIZE,
out bufLen
);
AL10.alGetBufferi(
buf.Handle,
AL10.AL_BITS,
out bits
);
bits /= 8;
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
byte[] data = new byte[bufLen];
byte[] monoData = new byte[bufLen / 2];
fixed(byte *dataPtr = &data[0])
fixed(byte *monoPtr = &monoData[0])
{
ALEXT.alGetBufferSamplesSOFT(
buf.Handle,
0,
bufLen / bits / 2,
ALEXT.AL_STEREO_SOFT,
bits == 2 ? ALEXT.AL_SHORT_SOFT : ALEXT.AL_BYTE_SOFT,
(IntPtr)dataPtr
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
if (bits == 2)
{
short *src = (short *)dataPtr;
short *dst = (short *)monoPtr;
for (int i = 0; i < monoData.Length / 2; i += 1)
{
dst[i] = (short)(((int)src[0] + (int)src[1]) / 2);
src += 2;
}
}
else
{
sbyte *src = (sbyte *)dataPtr;
sbyte *dst = (sbyte *)monoPtr;
for (int i = 0; i < monoData.Length; i += 1)
{
dst[i] = (sbyte)(((short)src[0] + (short)src[1]) / 2);
src += 2;
}
}
}
return(GenBuffer(
monoData,
(uint)buf.SampleRate,
1,
0,
0,
false,
(uint)bits - 1
));
}
