public SoundEffect(
byte[] buffer,
int offset,
int count,
int sampleRate,
AudioChannels channels,
int loopStart,
int loopLength
)
{
byte[] sendBuf;
if (offset != 0 || count != buffer.Length)
{
// I kind of hate this. -flibit
sendBuf = new byte[count];
Array.Copy(buffer, offset, sendBuf, 0, count);
}
else
{
sendBuf = buffer;
}
INTERNAL_buffer = AudioDevice.GenBuffer(
sendBuf,
(uint)sampleRate,
(uint)channels,
(uint)loopStart,
(uint)(loopStart + loopLength),
false,
1
);
}
public IALSource GenSource(IALBuffer buffer, bool isXACT)
{
uint result;
AL10.alGenSources(1, out result);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
if (result == 0)
{
return(null);
}
AL10.alSourcei(
result,
AL10.AL_BUFFER,
(int)(buffer as OpenALBuffer).Handle
);
AL10.alSourcef(
result,
AL10.AL_REFERENCE_DISTANCE,
AudioDevice.DistanceScale
);
if (isXACT)
{
AL10.alSourcef(
result,
AL10.AL_MAX_GAIN,
AudioDevice.MAX_GAIN_VALUE
);
}
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
return(new OpenALSource(result));
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int sampleRate
) {
// No-op, duh.
}
public void DeleteBuffer(IALBuffer buffer)
{
uint handle = (buffer as OpenALBuffer).Handle;
AL10.alDeleteBuffers(1, ref handle);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void SetBufferFloatData(
IALBuffer buffer,
IntPtr data,
int offset,
int count
)
{
// No-op, duh.
}
public void QueueSourceBuffer(IALSource source, IALBuffer buffer)
{
uint buf = (buffer as OpenALBuffer).Handle;
AL10.alSourceQueueBuffers(
(source as OpenALSource).Handle,
(IntPtr)1,
ref buf
);
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
IntPtr data,
int offset,
int count,
int sampleRate
)
{
// No-op, duh.
}
public void QueueSourceBuffer(IALSource source, IALBuffer buffer)
{
uint buf = (buffer as OpenALBuffer).Handle;
AL10.alSourceQueueBuffers(
(source as OpenALSource).Handle,
1,
ref buf
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void SubmitFloatBufferEXT(float[] buffer, int offset, int count)
{
/* 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)
{
availableBuffers.Enqueue(AudioDevice.ALDevice.GenBuffer());
}
// Push buffer to the AL.
IALBuffer newBuf = availableBuffers.Dequeue();
GCHandle handle = GCHandle.Alloc(buffer, GCHandleType.Pinned);
AudioDevice.ALDevice.SetBufferFloatData(
newBuf,
channels,
handle.AddrOfPinnedObject(),
offset,
count,
sampleRate
);
handle.Free();
// If we're already playing, queue immediately.
if (INTERNAL_alSource != null)
{
AudioDevice.ALDevice.QueueSourceBuffer(
INTERNAL_alSource,
newBuf
);
queuedBuffers.Enqueue(newBuf);
// If the source stopped, reboot it now.
if (AudioDevice.ALDevice.GetSourceState(INTERNAL_alSource) == SoundState.Stopped)
{
AudioDevice.ALDevice.PlaySource(INTERNAL_alSource);
}
}
else
{
buffersToQueue.Enqueue(newBuf);
}
PendingBufferCount += 1;
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[channels],
data,
(IntPtr)(data.Length * 4),
(IntPtr)sampleRate
);
}
public SoundEffect(
byte[] buffer,
int sampleRate,
AudioChannels channels
)
{
INTERNAL_buffer = AudioDevice.GenBuffer(
buffer,
(uint) sampleRate,
(uint) channels,
0,
0,
false,
1
);
}
public SoundEffect(
byte[] buffer,
int sampleRate,
AudioChannels channels
)
{
INTERNAL_buffer = AudioDevice.GenBuffer(
buffer,
(uint)sampleRate,
(uint)channels,
0,
0,
false,
1
);
}
public IALSource GenSource(IALBuffer buffer)
{
uint result;
AL10.alGenSources((IntPtr)1, out result);
if (result == 0)
{
return(null);
}
AL10.alSourcei(
result,
AL10.AL_BUFFER,
(int)(buffer as OpenALBuffer).Handle
);
return(new OpenALSource(result));
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
byte[] data,
int count,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToShort[channels],
data, // TODO: offset -flibit
(IntPtr)count,
(IntPtr)sampleRate
);
}
public void SubmitBuffer(byte[] buffer, int offset, int count)
{
// Generate a buffer if we don't have any to use.
if (availableBuffers.Count == 0)
{
availableBuffers.Enqueue(
AudioDevice.ALDevice.GenBuffer()
);
}
// Push the data to OpenAL.
IALBuffer newBuf = availableBuffers.Dequeue();
GCHandle handle = GCHandle.Alloc(buffer, GCHandleType.Pinned);
AudioDevice.ALDevice.SetBufferData(
newBuf,
channels,
handle.AddrOfPinnedObject(),
offset,
count,
sampleRate
);
handle.Free();
// If we're already playing, queue immediately.
if (INTERNAL_alSource != null)
{
AudioDevice.ALDevice.QueueSourceBuffer(
INTERNAL_alSource,
newBuf
);
queuedBuffers.Enqueue(newBuf);
// If the source stopped, reboot it now.
if (AudioDevice.ALDevice.GetSourceState(INTERNAL_alSource) == SoundState.Stopped)
{
AudioDevice.ALDevice.PlaySource(INTERNAL_alSource);
}
}
else
{
buffersToQueue.Enqueue(newBuf);
}
PendingBufferCount += 1;
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[channels],
data,
(IntPtr)(data.Length * 4),
(IntPtr)sampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
byte[] data,
int count,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToShort[channels],
data, // TODO: offset -flibit
(IntPtr)count,
(IntPtr)sampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void SetBufferFloatData(
IALBuffer buffer,
AudioChannels channels,
IntPtr data,
int offset,
int count,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[(int)channels],
data + (offset * 4),
count * 4,
sampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void SetBufferFloatData(
IALBuffer buffer,
IntPtr data,
int offset,
int count
)
{
OpenALBuffer buf = buffer as OpenALBuffer;
AL10.alBufferData(
buf.Handle,
XNAToFloat[buf.Channels],
data + (offset * 4),
count * 4,
buf.SampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public IALSource GenSource(IALBuffer buffer)
{
uint result;
AL10.alGenSources(1, out result);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
if (result == 0)
{
return(null);
}
AL10.alSourcei(
result,
AL10.AL_BUFFER,
(int)(buffer as OpenALBuffer).Handle
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
return(new OpenALSource(result));
}
/* 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)
{
availableBuffers.Enqueue(AudioDevice.ALDevice.GenBuffer());
}
// Push buffer to the AL.
IALBuffer newBuf = availableBuffers.Dequeue();
AudioDevice.ALDevice.SetBufferData(
newBuf,
channels,
buffer,
sampleRate
);
// If we're already playing, queue immediately.
if (INTERNAL_alSource != null)
{
AudioDevice.ALDevice.QueueSourceBuffer(
INTERNAL_alSource,
newBuf
);
queuedBuffers.Enqueue(newBuf);
}
else
{
buffersToQueue.Enqueue(newBuf);
}
PendingBufferCount += 1;
}
internal SoundEffect(
string name,
byte[] buffer,
uint sampleRate,
uint channels,
uint loopStart,
uint loopLength,
bool isADPCM,
uint formatParameter
)
{
Name = name;
INTERNAL_buffer = AudioDevice.GenBuffer(
buffer,
sampleRate,
channels,
loopStart,
loopStart + loopLength,
isADPCM,
formatParameter
);
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int offset,
int count,
int sampleRate
)
{
GCHandle handle = GCHandle.Alloc(data, GCHandleType.Pinned);
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[(int)channels],
handle.AddrOfPinnedObject() + (offset * 4),
(IntPtr)(count * 4),
(IntPtr)sampleRate
);
handle.Free();
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void SubmitBuffer(byte[] buffer, int offset, int count)
{
// Generate a buffer if we don't have any to use.
if (availableBuffers.Count == 0)
{
availableBuffers.Enqueue(
AudioDevice.ALDevice.GenBuffer()
);
}
// Push the data to OpenAL.
IALBuffer newBuf = availableBuffers.Dequeue();
AudioDevice.ALDevice.SetBufferData(
newBuf,
channels,
buffer, // TODO: offset -flibit
count,
sampleRate
);
// If we're already playing, queue immediately.
if (INTERNAL_alSource != null)
{
AudioDevice.ALDevice.QueueSourceBuffer(
INTERNAL_alSource,
newBuf
);
queuedBuffers.Enqueue(newBuf);
}
else
{
buffersToQueue.Enqueue(newBuf);
}
PendingBufferCount += 1;
}
public SoundEffect(
byte[] buffer,
int offset,
int count,
int sampleRate,
AudioChannels channels,
int loopStart,
int loopLength
)
{
byte[] sendBuf;
if (offset != 0 || count != buffer.Length)
{
// I kind of hate this. -flibit
sendBuf = new byte[count];
Array.Copy(buffer, offset, sendBuf, 0, count);
}
else
{
sendBuf = buffer;
}
INTERNAL_buffer = AudioDevice.GenBuffer(
sendBuf,
(uint) sampleRate,
(uint) channels,
(uint) loopStart,
(uint) (loopStart + loopLength),
false,
1
);
}
public IALSource GenSource(IALBuffer buffer)
{
uint result;
AL10.alGenSources((IntPtr) 1, out result);
if (result == 0)
{
return null;
}
AL10.alSourcei(
result,
AL10.AL_BUFFER,
(int) (buffer as OpenALBuffer).Handle
);
return new OpenALSource(result);
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[channels],
data,
(IntPtr) (data.Length * 4),
(IntPtr) sampleRate
);
}
public void SetBufferFloatData(
IALBuffer buffer,
AudioChannels channels,
IntPtr data,
int offset,
int count,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[(int) channels],
data + (offset * 4),
count * 4,
sampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void GetBufferData(
IALSource source,
IALBuffer[] buffer,
float[] samples,
AudioChannels channels
)
{
// No-op, duh.
}
public void DeleteBuffer(IALBuffer buffer)
{
uint handle = (buffer as OpenALBuffer).Handle;
AL10.alDeleteBuffers((IntPtr) 1, ref handle);
}
public void QueueSourceBuffer(IALSource source, IALBuffer buffer)
{
// No-op, duh.
}
public void QueueSourceBuffer(IALSource source, IALBuffer buffer)
{
uint buf = (buffer as OpenALBuffer).Handle;
AL10.alSourceQueueBuffers(
(source as OpenALSource).Handle,
(IntPtr) 1,
ref buf
);
}
private void INTERNAL_loadAudioStream(Stream s)
{
byte[] data;
uint sampleRate = 0;
uint numChannels = 0;
bool isADPCM = false;
uint formatParameter = 0;
using (BinaryReader reader = new BinaryReader(s))
{
// RIFF Signature
string signature = new string(reader.ReadChars(4));
if (signature != "RIFF")
{
throw new NotSupportedException("Specified stream is not a wave file.");
}
reader.ReadUInt32(); // Riff Chunk Size
string wformat = new string(reader.ReadChars(4));
if (wformat != "WAVE")
{
throw new NotSupportedException("Specified stream is not a wave file.");
}
// WAVE Header
string format_signature = new string(reader.ReadChars(4));
while (format_signature != "fmt ")
{
reader.ReadBytes(reader.ReadInt32());
format_signature = new string(reader.ReadChars(4));
}
int format_chunk_size = reader.ReadInt32();
// Header Information
uint audio_format = reader.ReadUInt16(); // 2
numChannels = reader.ReadUInt16(); // 4
sampleRate = reader.ReadUInt32(); // 8
reader.ReadUInt32(); // 12, Byte Rate
ushort blockAlign = reader.ReadUInt16(); // 14, Block Align
ushort bitDepth = reader.ReadUInt16(); // 16, Bits Per Sample
if (audio_format == 1)
{
System.Diagnostics.Debug.Assert(bitDepth == 8 || bitDepth == 16);
formatParameter = (uint)(bitDepth / 16); // 1 for 16, 0 for 8
}
else if (audio_format != 2)
{
isADPCM = true;
formatParameter = (((blockAlign / numChannels) - 6) * 2);
}
else
{
throw new NotSupportedException("Wave format is not supported.");
}
// Reads residual bytes
if (format_chunk_size > 16)
{
reader.ReadBytes(format_chunk_size - 16);
}
// data Signature
string data_signature = new string(reader.ReadChars(4));
while (data_signature.ToLowerInvariant() != "data")
{
reader.ReadBytes(reader.ReadInt32());
data_signature = new string(reader.ReadChars(4));
}
if (data_signature != "data")
{
throw new NotSupportedException("Specified wave file is not supported.");
}
int waveDataLength = reader.ReadInt32();
data = reader.ReadBytes(waveDataLength);
}
INTERNAL_buffer = AudioDevice.GenBuffer(
data,
sampleRate,
numChannels,
0,
0,
isADPCM,
formatParameter
);
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int offset,
int count,
int sampleRate
)
{
GCHandle handle = GCHandle.Alloc(data, GCHandleType.Pinned);
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[(int) channels],
handle.AddrOfPinnedObject() + (offset * 4),
(IntPtr) (count * 4),
(IntPtr) sampleRate
);
handle.Free();
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void GetBufferData(
IALSource source,
IALBuffer[] buffer,
IntPtr samples,
int samplesLen,
AudioChannels channels
) {
// No-op, duh.
}
public void QueueSourceBuffer(IALSource source, IALBuffer buffer)
{
// No-op, duh.
}
public IALSource GenSource(IALBuffer buffer)
{
return new NullSource();
}
internal SoundEffect(
string name,
byte[] buffer,
uint sampleRate,
uint channels,
uint loopStart,
uint loopLength,
bool isADPCM,
uint formatParameter
)
{
Name = name;
INTERNAL_buffer = AudioDevice.GenBuffer(
buffer,
sampleRate,
channels,
loopStart,
loopStart + loopLength,
isADPCM,
formatParameter
);
}
public IALSource GenSource(IALBuffer buffer)
{
return(new NullSource());
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
byte[] data,
int count,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToShort[channels],
data, // TODO: offset -flibit
(IntPtr) count,
(IntPtr) sampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
public void DeleteBuffer(IALBuffer buffer)
{
// No-op, duh.
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToFloat[channels],
data,
(IntPtr) (data.Length * 4),
(IntPtr) sampleRate
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
internal void Play(bool isManaged)
{
if (State != SoundState.Stopped)
{
return; // No-op if we're already playing.
}
if (INTERNAL_alSource != null)
{
// The sound has stopped, but hasn't cleaned up yet...
AudioDevice.ALDevice.StopAndDisposeSource(INTERNAL_alSource);
INTERNAL_alSource = null;
}
while (queuedBuffers.Count > 0)
{
availableBuffers.Enqueue(queuedBuffers.Dequeue());
PendingBufferCount -= 1;
}
INTERNAL_alSource = AudioDevice.ALDevice.GenSource();
if (INTERNAL_alSource == null)
{
FNAPlatform.Log("WARNING: AL SOURCE WAS NOT AVAILABLE. SKIPPING.");
return;
}
// Queue the buffers to this source
while (buffersToQueue.Count > 0)
{
IALBuffer nextBuf = buffersToQueue.Dequeue();
queuedBuffers.Enqueue(nextBuf);
AudioDevice.ALDevice.QueueSourceBuffer(INTERNAL_alSource, nextBuf);
}
// Apply Pan/Position
if (INTERNAL_positionalAudio)
{
INTERNAL_positionalAudio = false;
AudioDevice.ALDevice.SetSourcePosition(
INTERNAL_alSource,
position
);
}
else
{
Pan = Pan;
}
// Reassign Properties, in case the AL properties need to be applied.
Volume = Volume;
Pitch = Pitch;
// ... but wait! What if we need moar buffers?
for (
int i = MINIMUM_BUFFER_CHECK - PendingBufferCount;
(i > 0) && BufferNeeded != null;
i -= 1
)
{
BufferNeeded(this, null);
}
// Finally.
AudioDevice.ALDevice.PlaySource(INTERNAL_alSource);
if (isManaged)
{
AudioDevice.DynamicInstancePool.Add(this);
}
}
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
));
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
float[] data,
int sampleRate
)
{
// No-op, duh.
}
private void INTERNAL_loadAudioStream(Stream s)
{
byte[] data;
uint sampleRate = 0;
uint numChannels = 0;
bool isADPCM = false;
uint formatParameter = 0;
using (BinaryReader reader = new BinaryReader(s))
{
// RIFF Signature
string signature = new string(reader.ReadChars(4));
if (signature != "RIFF")
{
throw new NotSupportedException("Specified stream is not a wave file.");
}
reader.ReadUInt32(); // Riff Chunk Size
string wformat = new string(reader.ReadChars(4));
if (wformat != "WAVE")
{
throw new NotSupportedException("Specified stream is not a wave file.");
}
// WAVE Header
string format_signature = new string(reader.ReadChars(4));
while (format_signature != "fmt ")
{
reader.ReadBytes(reader.ReadInt32());
format_signature = new string(reader.ReadChars(4));
}
int format_chunk_size = reader.ReadInt32();
// Header Information
uint audio_format = reader.ReadUInt16(); // 2
numChannels = reader.ReadUInt16(); // 4
sampleRate = reader.ReadUInt32(); // 8
reader.ReadUInt32(); // 12, Byte Rate
ushort blockAlign = reader.ReadUInt16(); // 14, Block Align
ushort bitDepth = reader.ReadUInt16(); // 16, Bits Per Sample
if (audio_format == 1)
{
formatParameter = (uint) (bitDepth / 16); // 1 for 16, 0 for 8
}
else if (audio_format != 2)
{
isADPCM = true;
formatParameter = (((blockAlign / numChannels) - 6) * 2);
}
else
{
throw new NotSupportedException("Wave format is not supported.");
}
// Reads residual bytes
if (format_chunk_size > 16)
{
reader.ReadBytes(format_chunk_size - 16);
}
// data Signature
string data_signature = new string(reader.ReadChars(4));
while (data_signature.ToLowerInvariant() != "data")
{
reader.ReadBytes(reader.ReadInt32());
data_signature = new string(reader.ReadChars(4));
}
if (data_signature != "data")
{
throw new NotSupportedException("Specified wave file is not supported.");
}
int waveDataLength = reader.ReadInt32();
data = reader.ReadBytes(waveDataLength);
}
INTERNAL_buffer = AudioDevice.GenBuffer(
data,
sampleRate,
numChannels,
0,
0,
isADPCM,
formatParameter
);
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
IntPtr data,
int offset,
int count,
int sampleRate
) {
// No-op, duh.
}
public void SetBufferData(
IALBuffer buffer,
AudioChannels channels,
byte[] data,
int count,
int sampleRate
)
{
AL10.alBufferData(
(buffer as OpenALBuffer).Handle,
XNAToShort[channels],
data, // TODO: offset -flibit
(IntPtr) count,
(IntPtr) sampleRate
);
}
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 void DeleteBuffer(IALBuffer buffer)
{
uint handle = (buffer as OpenALBuffer).Handle;
AL10.alDeleteBuffers((IntPtr) 1, ref handle);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
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 IALSource GenSource(IALBuffer buffer)
{
uint result;
AL10.alGenSources((IntPtr) 1, out result);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
if (result == 0)
{
return null;
}
AL10.alSourcei(
result,
AL10.AL_BUFFER,
(int) (buffer as OpenALBuffer).Handle
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
return new OpenALSource(result);
}
public void DeleteBuffer(IALBuffer buffer)
{
// No-op, duh.
}
public void QueueSourceBuffer(IALSource source, IALBuffer buffer)
{
uint buf = (buffer as OpenALBuffer).Handle;
AL10.alSourceQueueBuffers(
(source as OpenALSource).Handle,
(IntPtr) 1,
ref buf
);
#if VERBOSE_AL_DEBUGGING
CheckALError();
#endif
}
