public IALBuffer GenBuffer()
{
uint result;
AL10.alGenBuffers((IntPtr)1, out result);
return(new OpenALBuffer(result, TimeSpan.Zero));
}
protected AudioBuffer(int samplingRate)
{
SamplingRate = samplingRate;
AL10.alGenBuffers(1, out Buffer);
Check();
LastUpdatedDateTime = DateTime.Now;
}
public SoundBuffer()
{
AL10.alGenBuffers(1, out _handle);
ALUtils.CheckALError("unable to create sound buffer");
if (_handle == 0)
{
throw new AudioException("Unable to create OpenAL sound buffer");
}
}
public IALBuffer GenBuffer(int sampleRate, AudioChannels channels)
{
uint result;
AL10.alGenBuffers(1, out result);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
return(new OpenALBuffer(result, TimeSpan.Zero, (int)channels, sampleRate));
}
public IALBuffer GenBuffer()
{
uint result;
AL10.alGenBuffers(1, out result);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
return(new OpenALBuffer(result, TimeSpan.Zero));
}
/// <summary>
/// Create a number of OpenAL buffers for this device.
/// </summary>
/// <param name="buffers">Array to fill with buffer names.</param>
/// <param name="n">Number of buffers to generate.</param>
public void CreateBuffers(uint[] buffers, int n)
{
CheckDisposed();
ALC10.alcMakeContextCurrent(MainContext.Handle);
AL10.alGenBuffers(n, buffers);
AlHelper.AlAlwaysCheckError("Failed to generate buffers.");
for (var i = 0; i < n; i++)
{
_buffers.Add(buffers[i]);
}
}
/// <summary>
/// Create an OpenAL buffer for this device.
/// </summary>
public uint CreateBuffer()
{
CheckDisposed();
ALC10.alcMakeContextCurrent(MainContext.Handle);
AL10.alGenBuffers(1, out var name);
if (name == 0)
{
AlHelper.AlCheckError("alGenBuffer call failed.");
throw new Exception("Failed to create buffer.");
}
_buffers.Add(name);
return(name);
}
/* THIS IS AN EXTENSION OF THE XNA4 API! */
public void SubmitFloatBufferEXT(float[] buffer)
{
/* Float samples are the typical format received from decoders.
* We currently use this for the VideoPlayer.
* -flibit
*/
// Generate a buffer if we don't have any to use.
if (availableBuffers.Count == 0)
{
uint buf;
AL10.alGenBuffers((IntPtr)1, out buf);
availableBuffers.Enqueue(buf);
}
// Push the data to OpenAL.
uint newBuf = availableBuffers.Dequeue();
AL10.alBufferData(
newBuf,
XNAToFloat[channels],
buffer,
(IntPtr)(buffer.Length * 4),
(IntPtr)sampleRate
);
// If we're already playing, queue immediately.
if (State == SoundState.Playing)
{
AL10.alSourceQueueBuffers(
INTERNAL_alSource,
(IntPtr)1,
ref newBuf
);
queuedBuffers.Enqueue(newBuf);
}
else
{
buffersToQueue.Enqueue(newBuf);
}
PendingBufferCount += 1;
}
public void SubmitBuffer(byte[] buffer, int offset, int count)
{
// Generate a buffer if we don't have any to use.
if (availableBuffers.Count == 0)
{
uint buf;
AL10.alGenBuffers((IntPtr)1, out buf);
availableBuffers.Enqueue(buf);
}
// Push the data to OpenAL.
uint newBuf = availableBuffers.Dequeue();
AL10.alBufferData(
newBuf,
XNAToShort[channels],
buffer, // TODO: offset -flibit
(IntPtr)count,
(IntPtr)sampleRate
);
// If we're already playing, queue immediately.
if (State == SoundState.Playing)
{
AL10.alSourceQueueBuffers(
INTERNAL_alSource,
(IntPtr)1,
ref newBuf
);
queuedBuffers.Enqueue(newBuf);
}
else
{
buffersToQueue.Enqueue(newBuf);
}
PendingBufferCount += 1;
}
public IALBuffer GenBuffer(
byte[] data,
uint sampleRate,
uint channels,
uint loopStart,
uint loopEnd,
bool isADPCM,
uint formatParameter
)
{
uint result;
// Generate the buffer now, in case we need to perform alBuffer ops.
AL10.alGenBuffers(1, out result);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
int format;
int length = data.Length;
if (isADPCM)
{
format = (channels == 2) ?
ALEXT.AL_FORMAT_STEREO_MSADPCM_SOFT :
ALEXT.AL_FORMAT_MONO_MSADPCM_SOFT;
AL10.alBufferi(
result,
ALEXT.AL_UNPACK_BLOCK_ALIGNMENT_SOFT,
(int)formatParameter
);
}
else
{
if (formatParameter == 1)
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO16:
AL10.AL_FORMAT_MONO16;
/* We have to perform extra data validation on
* PCM16 data, as the MS SoundEffect builder will
* leave extra bytes at the end which will confuse
* alBufferData and throw an AL_INVALID_VALUE.
* -flibit
*/
length &= 0x7FFFFFFE;
}
else
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO8:
AL10.AL_FORMAT_MONO8;
}
}
// Load it!
AL10.alBufferData(
result,
format,
data,
length,
(int)sampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
// Calculate the duration now, after we've unpacked the buffer
int bufLen, bits;
AL10.alGetBufferi(
result,
AL10.AL_SIZE,
out bufLen
);
AL10.alGetBufferi(
result,
AL10.AL_BITS,
out bits
);
if (bufLen == 0 || bits == 0)
{
throw new InvalidOperationException(
"OpenAL buffer allocation failed!"
);
}
TimeSpan resultDur = TimeSpan.FromSeconds(
bufLen /
(bits / 8) /
channels /
((double)sampleRate)
);
// Set the loop points, if applicable
if (loopStart > 0 || loopEnd > 0)
{
AL10.alBufferiv(
result,
ALEXT.AL_LOOP_POINTS_SOFT,
new int[]
{
(int)loopStart,
(int)loopEnd
}
);
}
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
// Finally.
return(new OpenALBuffer(result, resultDur, (int)channels, (int)sampleRate));
}
private void INTERNAL_bufferData(
byte[] data,
uint sampleRate,
uint channels,
uint loopStart,
uint loopEnd,
bool isADPCM,
uint formatParameter
)
{
if (OpenALDevice.Instance == null)
{
throw new NoAudioHardwareException();
}
// Generate the buffer now, in case we need to perform alBuffer ops.
AL10.alGenBuffers((IntPtr)1, out INTERNAL_buffer);
int format;
if (isADPCM)
{
format = (channels == 2) ?
ALEXT.AL_FORMAT_STEREO_MSADPCM_SOFT :
ALEXT.AL_FORMAT_MONO_MSADPCM_SOFT;
AL10.alBufferi(
INTERNAL_buffer,
ALEXT.AL_UNPACK_BLOCK_ALIGNMENT_SOFT,
(int)formatParameter
);
}
else
{
if (formatParameter == 1)
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO16:
AL10.AL_FORMAT_MONO16;
}
else
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO8:
AL10.AL_FORMAT_MONO8;
}
}
// Load it!
AL10.alBufferData(
INTERNAL_buffer,
format,
data,
(IntPtr)data.Length,
(IntPtr)sampleRate
);
// Calculate the duration now, after we've unpacked the buffer
int bufLen, bits;
AL10.alGetBufferi(
INTERNAL_buffer,
AL10.AL_SIZE,
out bufLen
);
AL10.alGetBufferi(
INTERNAL_buffer,
AL10.AL_BITS,
out bits
);
Duration = TimeSpan.FromSeconds(
bufLen /
(bits / 8) /
channels /
((double)sampleRate)
);
// Set the loop points, if applicable
if (loopStart > 0 || loopEnd > 0)
{
AL10.alBufferiv(
INTERNAL_buffer,
ALEXT.AL_LOOP_POINTS_SOFT,
new int[]
{
(int)loopStart,
(int)loopEnd
}
);
}
}
public AudioBuffer()
{
AL10.alGenBuffers(1, out _handle);
AudioSystem.alCheckError();
}
public OpenAlSoundSource(byte[] data, int byteCount, int channels, int sampleBits, int sampleRate)
{
SampleRate = sampleRate;
AL10.alGenBuffers(1, out buffer);
AL10.alBufferData(buffer, OpenAlSoundEngine.MakeALFormat(channels, sampleBits), data, byteCount, sampleRate);
}
public OpenAlSoundSource(byte[] data, int channels, int sampleBits, int sampleRate)
{
AL10.alGenBuffers(new IntPtr(1), out Buffer);
AL10.alBufferData(Buffer, MakeALFormat(channels, sampleBits), data, new IntPtr(data.Length), new IntPtr(sampleRate));
}
public IALBuffer GenBuffer(
byte[] data,
uint sampleRate,
uint channels,
uint loopStart,
uint loopEnd,
bool isADPCM,
uint formatParameter
)
{
uint result;
// Generate the buffer now, in case we need to perform alBuffer ops.
AL10.alGenBuffers((IntPtr)1, out result);
int format;
if (isADPCM)
{
format = (channels == 2) ?
ALEXT.AL_FORMAT_STEREO_MSADPCM_SOFT :
ALEXT.AL_FORMAT_MONO_MSADPCM_SOFT;
AL10.alBufferi(
result,
ALEXT.AL_UNPACK_BLOCK_ALIGNMENT_SOFT,
(int)formatParameter
);
}
else
{
if (formatParameter == 1)
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO16:
AL10.AL_FORMAT_MONO16;
}
else
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO8:
AL10.AL_FORMAT_MONO8;
}
}
// Load it!
AL10.alBufferData(
result,
format,
data,
(IntPtr)data.Length,
(IntPtr)sampleRate
);
// Calculate the duration now, after we've unpacked the buffer
int bufLen, bits;
AL10.alGetBufferi(
result,
AL10.AL_SIZE,
out bufLen
);
AL10.alGetBufferi(
result,
AL10.AL_BITS,
out bits
);
TimeSpan resultDur = TimeSpan.FromSeconds(
bufLen /
(bits / 8) /
channels /
((double)sampleRate)
);
// Set the loop points, if applicable
if (loopStart > 0 || loopEnd > 0)
{
AL10.alBufferiv(
result,
ALEXT.AL_LOOP_POINTS_SOFT,
new int[]
{
(int)loopStart,
(int)loopEnd
}
);
}
// Finally.
return(new OpenALBuffer(result, resultDur));
}
private void INTERNAL_bufferData(
byte[] data,
uint sampleRate,
uint channels,
uint loopStart,
uint loopEnd,
bool isADPCM,
uint formatParameter
)
{
if (OpenALDevice.Instance == null)
{
throw new NoAudioHardwareException();
}
// Generate the buffer now, in case we need to perform alBuffer ops.
AL10.alGenBuffers((IntPtr)1, out INTERNAL_buffer);
int format;
if (isADPCM)
{
Platform.AssertSupported("ADPCM");
format = (channels == 2) ?
ALEXT.AL_FORMAT_STEREO_MSADPCM_SOFT :
ALEXT.AL_FORMAT_MONO_MSADPCM_SOFT;
AL10.alBufferi(
INTERNAL_buffer,
ALEXT.AL_UNPACK_BLOCK_ALIGNMENT_SOFT,
(int)formatParameter
);
}
else
{
if (formatParameter == 1)
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO16:
AL10.AL_FORMAT_MONO16;
}
else
{
format = (channels == 2) ?
AL10.AL_FORMAT_STEREO8:
AL10.AL_FORMAT_MONO8;
}
}
// Load it!
AL10.alBufferData(
INTERNAL_buffer,
format,
data,
(IntPtr)data.Length,
(IntPtr)sampleRate
);
// Calculate the duration now, after we've unpacked the buffer
int bufLen, bits;
AL10.alGetBufferi(
INTERNAL_buffer,
AL10.AL_SIZE,
out bufLen
);
AL10.alGetBufferi(
INTERNAL_buffer,
AL10.AL_BITS,
out bits
);
Duration = TimeSpan.FromSeconds(
bufLen /
(bits / 8) /
channels /
((double)sampleRate)
);
// Compute the default loop end point (end of the buffer), because
// some content builders automatically set a loop endpoint here instead of at 0
int defaultEndPoint = (bufLen / (bits / 8)) / (int)channels;
var hasCustomStartPoint = (loopStart > 0);
var hasCustomEndPoint = (
(loopEnd > loopStart) &&
(loopEnd < defaultEndPoint)
);
if (hasCustomStartPoint || hasCustomEndPoint)
{
// Set the loop points, if applicable
Platform.AssertSupported("CustomLoopPoints");
AL10.alBufferiv(
INTERNAL_buffer,
ALEXT.AL_LOOP_POINTS_SOFT,
new int[]
{
(int)loopStart,
(int)loopEnd
}
);
}
}