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CWE-ou-minkata/Sources/Plasma/PubUtilLib/plAudio/plDSoundBuffer.cpp

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/*==LICENSE==*
CyanWorlds.com Engine - MMOG client, server and tools
Copyright (C) 2011 Cyan Worlds, Inc.
This program is free software: you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation, either version 3 of the License, or
(at your option) any later version.
This program is distributed in the hope that it will be useful,
but WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
GNU General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program. If not, see <http://www.gnu.org/licenses/>.
You can contact Cyan Worlds, Inc. by email legal@cyan.com
or by snail mail at:
Cyan Worlds, Inc.
14617 N Newport Hwy
Mead, WA 99021
*==LICENSE==*/
//////////////////////////////////////////////////////////////////////////////
// //
// plDSoundBuffer - Simple wrapper class for a DirectSound buffer. //
// Allows us to simplify all the work done behind the //
// scenes in plWin32BufferThread. //
// //
//////////////////////////////////////////////////////////////////////////////
#include "hsTypes.h"
#include "hsThread.h"
#include "plDSoundBuffer.h"
#include <al.h>
#include "plgDispatch.h"
#include "plAudioSystem.h"
#include "plAudioCore/plAudioCore.h"
#include "plAudioCore/plAudioFileReader.h"
#include "plEAXEffects.h"
#include "plProfile.h"
#include "plStatusLog/plStatusLog.h"
UInt32 plDSoundBuffer::fNumBuffers = 0;
plProfile_CreateCounterNoReset( "Playing", "Sound", SoundPlaying );
plProfile_CreateCounterNoReset( "Allocated", "Sound", NumAllocated );
//// Constructor/Destructor //////////////////////////////////////////////////
plDSoundBuffer::plDSoundBuffer( UInt32 size, plWAVHeader &bufferDesc, hsBool enable3D, hsBool isLooping, hsBool tryStatic, bool streaming )
{
fLooping = isLooping;
fValid = false;
fBufferDesc = nil;
fLockPtr = nil;
fLockLength = 0;
fStreaming = streaming;
buffer = 0;
source = 0;
for(int i = 0; i < STREAMING_BUFFERS; ++i)
{
streamingBuffers[i] = 0;
}
IAllocate( size, bufferDesc, enable3D, tryStatic );
fNumBuffers++;
}
plDSoundBuffer::~plDSoundBuffer()
{
IRelease();
fNumBuffers--;
}
//// IAllocate ///////////////////////////////////////////////////////////////
void plDSoundBuffer::IAllocate( UInt32 size, plWAVHeader &bufferDesc, hsBool enable3D, hsBool tryStatic )
{
// Create a DSound buffer description
fBufferDesc = TRACKED_NEW plWAVHeader;
*fBufferDesc = bufferDesc;
fBufferSize = size;
// Do we want to try EAX?
if( plgAudioSys::UsingEAX() )
fEAXSource.Init( this );
fValid = true;
plProfile_Inc( NumAllocated );
}
//// IRelease ////////////////////////////////////////////////////////////////
void plDSoundBuffer::IRelease( void )
{
if( IsPlaying() )
Stop();
// Release stuff
fEAXSource.Release();
alSourcei(source, AL_BUFFER, nil);
alDeleteSources(1, &source);
if(buffer)
alDeleteBuffers( 1, &buffer );
else
alDeleteBuffers(STREAMING_BUFFERS, streamingBuffers);
source = 0;
buffer = 0;
alGetError();
memset(streamingBuffers, 0, STREAMING_BUFFERS * sizeof(unsigned));
delete fBufferDesc;
fBufferDesc = nil;
fBufferSize = 0;
fValid = false;
plProfile_Dec( NumAllocated );
}
/*****************************************************************************
*
* OpenAL
*
***/
int plDSoundBuffer::IGetALFormat(unsigned bitsPerSample, unsigned int numChannels)
{
int format = 0;
switch(bitsPerSample)
{
case 8:
format = (numChannels == 1) ? AL_FORMAT_MONO8 : AL_FORMAT_STEREO8;
break;
case 16:
format = (numChannels == 1) ? AL_FORMAT_MONO16 : AL_FORMAT_STEREO16;
break;
}
return format;
}
bool plDSoundBuffer::FillBuffer(void *data, unsigned bytes, plWAVHeader *header)
{
if(source)
{
alSourcei(source, AL_BUFFER, nil);
alDeleteSources(1, &source);
}
if(buffer)
alDeleteBuffers(1, &buffer);
source = 0;
buffer = 0;
ALenum format = IGetALFormat(fBufferDesc->fBitsPerSample, fBufferDesc->fNumChannels);
ALenum error = alGetError();
alGenBuffers(1, &buffer);
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create sound buffer %d", error);
return false;
}
alBufferData(buffer, format, data, bytes, header->fNumSamplesPerSec );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to fill sound buffer %d", error);
return false;
}
alGenSources(1, &source);
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create audio source %d %d", error, source);
return false;
}
// Just make it quiet for now
SetScalarVolume(0);
alSourcef(source, AL_ROLLOFF_FACTOR, 0.3048);
alGetError();
if( error != AL_NO_ERROR )
{
return false;
}
alSourcei(source, AL_BUFFER, buffer);
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to attach buffer to source %d", error);
return false;
}
return true;
}
//============================================================================
// OpenAL Streaming functions
//============================================================================
// this function is used when restarting the audio system. It is needed to restart a streaming source from where it left off
bool plDSoundBuffer::SetupStreamingSource(plAudioFileReader *stream)
{
unsigned char data[STREAM_BUFFER_SIZE];
unsigned int size;
ALenum error;
alGetError();
int numBuffersToQueue = 0;
// fill buffers with data
for( int i = 0; i < STREAMING_BUFFERS; i++ )
{
size = stream->NumBytesLeft() < STREAM_BUFFER_SIZE ? stream->NumBytesLeft() : STREAM_BUFFER_SIZE;
if(!size)
{
if(IsLooping())
{
stream->SetPosition(0);
}
}
stream->Read(size, data);
numBuffersToQueue++;
alGenBuffers( 1, &streamingBuffers[i] );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create sound buffer %d", error);
return false;
}
ALenum format = IGetALFormat(fBufferDesc->fBitsPerSample, fBufferDesc->fNumChannels);
alBufferData( streamingBuffers[i], format, data, size, fBufferDesc->fNumSamplesPerSec );
if( (error = alGetError()) != AL_NO_ERROR )
plStatusLog::AddLineS("audio.log", "alBufferData");
}
// Generate AL Source
alGenSources( 1, &source );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create audio source %d %d", error, source);
return false;
}
alSourcei(source, AL_BUFFER, nil);
SetScalarVolume(0);
alSourcef(source, AL_ROLLOFF_FACTOR, 0.3048);
alGetError();
if( error != AL_NO_ERROR )
{
return false;
}
alSourceQueueBuffers( source, numBuffersToQueue, streamingBuffers );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to queue buffers %d", error);
return false;
}
return true;
}
// this function is used when starting up a streaming sound, as opposed to restarting it due to an audio system restart.
bool plDSoundBuffer::SetupStreamingSource(void *data, unsigned bytes)
{
unsigned char bufferData[STREAM_BUFFER_SIZE];
unsigned int size;
ALenum error;
char *pData = (char *)data;
alGetError();
int numBuffersToQueue = 0;
// fill buffers with data
for( int i = 0; i < STREAMING_BUFFERS; i++ )
{
size = bytes < STREAM_BUFFER_SIZE ? bytes : STREAM_BUFFER_SIZE;
if(!size)
break;
MemCopy(bufferData, pData, size);
pData += size;
bytes-= size;
numBuffersToQueue++;
alGenBuffers( 1, &streamingBuffers[i] );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create sound buffer %d", error);
return false;
}
ALenum format = IGetALFormat(fBufferDesc->fBitsPerSample, fBufferDesc->fNumChannels);
alBufferData( streamingBuffers[i], format, bufferData, size, fBufferDesc->fNumSamplesPerSec );
if( (error = alGetError()) != AL_NO_ERROR )
plStatusLog::AddLineS("audio.log", "alBufferData");
}
// Generate AL Source
alGenSources( 1, &source );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create audio source %d %d", error, source);
return false;
}
alSourcei(source, AL_BUFFER, nil);
SetScalarVolume(0);
alSourcef(source, AL_ROLLOFF_FACTOR, 0.3048);
alGetError();
if( error != AL_NO_ERROR )
{
return false;
}
alSourceQueueBuffers( source, numBuffersToQueue, streamingBuffers );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to queue buffers %d", error);
return false;
}
return true;
}
//============================================================================
int plDSoundBuffer::BuffersProcessed()
{
if(alIsSource(source)==AL_FALSE)
{
plStatusLog::AddLineS("audio.log", "BuffersProcessed, source invalid");
return 0;
}
ALint processed = 0;
alGetSourcei( source, AL_BUFFERS_PROCESSED, &processed );
if(alGetError() != AL_NO_ERROR)
{
plStatusLog::AddLineS("audio.log", "alGetSourcei failed");
}
return processed;
}
//============================================================================
int plDSoundBuffer::BuffersQueued()
{
if(alIsSource(source)==AL_FALSE) return 0;
ALint queued = 0;
alGetSourcei( source, AL_BUFFERS_QUEUED, &queued );
alGetError();
return queued;
}
//============================================================================
bool plDSoundBuffer::StreamingFillBuffer(plAudioFileReader *stream)
{
if(!source)
return false;
ALenum error;
ALuint bufferId;
unsigned char data[STREAM_BUFFER_SIZE];
int buffersProcessed = BuffersProcessed();
hsBool finished = false;
for(int i = 0; i < buffersProcessed; i++)
{
alSourceUnqueueBuffers( source, 1, &bufferId );
if( (error = alGetError()) != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to unqueue buffer %d", error);
return false;
}
if(!finished)
{
if(stream->NumBytesLeft() == 0)
{
// if at anytime we run out of data, and we are looping, reset the data stream and continue to fill buffers
if(IsLooping())
{
stream->SetPosition(0); // we are looping, so reset data stream, and keep filling buffers
}
else
{
finished = true; // no more data, but we could still be playing, so we don't want to stop the sound yet
}
}
if(!finished)
{ unsigned int size = stream->NumBytesLeft() < STREAM_BUFFER_SIZE ? stream->NumBytesLeft() : STREAM_BUFFER_SIZE;
stream->Read(size, data);
ALenum format = IGetALFormat(fBufferDesc->fBitsPerSample, fBufferDesc->fNumChannels);
alBufferData( bufferId, format, data, size, fBufferDesc->fNumSamplesPerSec );
if( (error = alGetError()) != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to copy data to sound buffer %d", error);
return false;
}
alSourceQueueBuffers( source, 1, &bufferId );
if( (error = alGetError()) != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to queue buffer %d", error);
return false;
}
}
}
}
if(!IsPlaying() && !finished)
{
alSourcePlay(source);
}
alGetError();
return true;
}
/*****************************************************************************
*
* Voice playback functions
*
***/
bool plDSoundBuffer::GetAvailableBufferId(unsigned *bufferId)
{
if(mAvailableBuffers.empty())
{
return false;
}
*bufferId = mAvailableBuffers.front();
mAvailableBuffers.pop_front();
return true;
}
bool plDSoundBuffer::SetupVoiceSource()
{
ALenum error;
alGetError();
// Generate AL Buffers
alGenBuffers( STREAMING_BUFFERS, streamingBuffers );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create sound buffer %d", error);
return false;
}
for( int i = 0; i < STREAMING_BUFFERS; i++ )
{
mAvailableBuffers.push_back(streamingBuffers[i]);
}
// Generate AL Source
alGenSources( 1, &source );
error = alGetError();
if( error != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to create audio source %d %d", error, source);
return false;
}
SetScalarVolume(0);
alSourcef(source, AL_ROLLOFF_FACTOR, 0.3048);
alGetError();
if( error != AL_NO_ERROR )
{
return false;
}
alSourcei(source, AL_BUFFER, nil);
alGetError();
//alSourcei(source, AL_PITCH, 0);
// dont queue any buffers here
return true;
}
//============================================================================
void plDSoundBuffer::UnQueueVoiceBuffers()
{
unsigned buffersProcessed = BuffersProcessed();
if(buffersProcessed)
plStatusLog::AddLineS("audio.log", "unqueuing buffers %d", buffersProcessed);
for(int i = 0; i < buffersProcessed; i++)
{
ALuint unQueued;
alSourceUnqueueBuffers( source, 1, &unQueued );
if(alGetError() == AL_NO_ERROR)
{
mAvailableBuffers.push_back(unQueued);
}
else
{
plStatusLog::AddLineS("audio.log", "Failed to unqueue buffer");
}
}
}
//============================================================================
bool plDSoundBuffer::VoiceFillBuffer(void *data, unsigned bytes, unsigned bufferId)
{
if(!source)
return false;
ALenum error;
unsigned int size = bytes < STREAM_BUFFER_SIZE ? bytes : STREAM_BUFFER_SIZE;
ALenum format = IGetALFormat(fBufferDesc->fBitsPerSample, fBufferDesc->fNumChannels);
alBufferData( bufferId, format, data, size, fBufferDesc->fNumSamplesPerSec );
if( (error = alGetError()) != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to copy data to sound buffer %d", error);
return false;
}
alSourceQueueBuffers( source, 1, &bufferId );
if( (error = alGetError()) != AL_NO_ERROR )
{
plStatusLog::AddLineS("audio.log", "Failed to queue buffer %d", error);
return false;
}
if(!IsPlaying())
{
alSourcePlay(source);
}
alGetError();
return true;
}
//// SetLooping //////////////////////////////////////////////////////////////
void plDSoundBuffer::SetLooping( hsBool loop )
{
fLooping = loop;
}
void plDSoundBuffer::SetMinDistance( int dist )
{
alSourcei(source, AL_REFERENCE_DISTANCE, dist);
ALenum error;
if((error = alGetError()) != AL_NO_ERROR)
plStatusLog::AddLineS("audio.log", "Failed to set min distance");
}
void plDSoundBuffer::SetMaxDistance( int dist )
{
alSourcei(source, AL_MAX_DISTANCE, dist);
ALenum error;
if((error = alGetError()) != AL_NO_ERROR)
plStatusLog::AddLineS("audio.log", "Failed to set min distance");
}
//// Play ////////////////////////////////////////////////////////////////////
void plDSoundBuffer::Play( void )
{
if(!source)
return;
ALenum error = alGetError(); // clear error
// we dont want openal to loop our streaming buffers, or the buffer will loop back on itself. We will handle looping in the streaming sound
if(fLooping && !fStreaming)
alSourcei(source, AL_LOOPING, AL_TRUE);
else
alSourcei(source, AL_LOOPING, AL_FALSE);
error = alGetError();
alSourcePlay(source);
error = alGetError();
if(error != AL_NO_ERROR)
plStatusLog::AddLineS("voice.log", "Play failed");
plProfile_Inc( SoundPlaying );
}
//// Stop ////////////////////////////////////////////////////////////////////
void plDSoundBuffer::Stop( void )
{
if(!source)
return;
alSourceStop(source);
alGetError();
plProfile_Dec( SoundPlaying );
}
//============================================================================
void plDSoundBuffer::SetPosition(float x, float y, float z)
{
alSource3f(source, AL_POSITION, x, y, -z); // negate z coord, since openal uses opposite handedness
alGetError();
}
//============================================================================
void plDSoundBuffer::SetOrientation(float x, float y, float z)
{
alSource3f(source, AL_ORIENTATION, x, y, -z); // negate z coord, since openal uses opposite handedness
alGetError();
}
//============================================================================
void plDSoundBuffer::SetVelocity(float x, float y, float z)
{
alSource3f(source, AL_VELOCITY, 0, 0, 0); // no doppler shift
alGetError();
}
//============================================================================
void plDSoundBuffer::SetConeAngles(int inner, int outer)
{
alSourcei(source, AL_CONE_INNER_ANGLE, inner);
alSourcei(source, AL_CONE_OUTER_ANGLE, outer);
alGetError();
}
//============================================================================
void plDSoundBuffer::SetConeOrientation(float x, float y, float z)
{
alSource3f(source, AL_DIRECTION, x, y, -z); // negate z coord, since openal uses opposite handedness
alGetError();
}
//============================================================================
// vol range: -5000 - 0
void plDSoundBuffer::SetConeOutsideVolume(int vol)
{
float volume = (float)vol / 5000.0f + 1.0f; // mb to scalar
alSourcef(source, AL_CONE_OUTER_GAIN, volume);
alGetError();
}
//============================================================================
void plDSoundBuffer::Rewind()
{
alSourceRewind(source);
alGetError();
}
//// IsPlaying ///////////////////////////////////////////////////////////////
hsBool plDSoundBuffer::IsPlaying( void )
{
ALint state = AL_STOPPED;
alGetSourcei(source, AL_SOURCE_STATE, &state);
alGetError();
return state == AL_PLAYING;
}
//// IsEAXAccelerated ////////////////////////////////////////////////////////
hsBool plDSoundBuffer::IsEAXAccelerated( void ) const
{
return fEAXSource.IsValid();
}
//// BytePosToMSecs //////////////////////////////////////////////////////////
UInt32 plDSoundBuffer::BytePosToMSecs( UInt32 bytePos ) const
{
return (UInt32)(bytePos * 1000 / (hsScalar)fBufferDesc->fAvgBytesPerSec);
}
//// GetBufferBytePos ////////////////////////////////////////////////////////
UInt32 plDSoundBuffer::GetBufferBytePos( hsScalar timeInSecs ) const
{
hsAssert( fBufferDesc != nil, "Nil buffer description when calling GetBufferBytePos()" );
UInt32 byte = (UInt32)( timeInSecs * (hsScalar)fBufferDesc->fNumSamplesPerSec );
byte *= fBufferDesc->fBlockAlign;
return byte;
}
//// GetLengthInBytes ////////////////////////////////////////////////////////
UInt32 plDSoundBuffer::GetLengthInBytes( void ) const
{
return fBufferSize;
}
//// SetEAXSettings //////////////////////////////////////////////////////////
void plDSoundBuffer::SetEAXSettings( plEAXSourceSettings *settings, hsBool force )
{
fEAXSource.SetFrom( settings, source, force );
}
//// GetBlockAlign ///////////////////////////////////////////////////////////
UInt8 plDSoundBuffer::GetBlockAlign( void ) const
{
return ( fBufferDesc != nil ) ? fBufferDesc->fBlockAlign : 0;
}
//// SetScalarVolume /////////////////////////////////////////////////////////
// Sets the volume, but on a range from 0 to 1
void plDSoundBuffer::SetScalarVolume( hsScalar volume )
{
if(source)
{
ALenum error;
alSourcef(source, AL_GAIN, volume);
if((error = alGetError()) != AL_NO_ERROR)
plStatusLog::AddLineS("audio.log", "failed to set volume on source %d", error);
}
}
unsigned plDSoundBuffer::GetByteOffset()
{
ALint bytes;
alGetSourcei(source, AL_BYTE_OFFSET, &bytes);
ALenum error = alGetError();
return bytes;
}
float plDSoundBuffer::GetTimeOffsetSec()
{
float time;
alGetSourcef(source, AL_SEC_OFFSET, &time);
ALenum error = alGetError();
return time;
}
void plDSoundBuffer::SetTimeOffsetSec(float seconds)
{
alSourcef(source, AL_SEC_OFFSET, seconds);
ALenum error = alGetError();
}
void plDSoundBuffer::SetTimeOffsetBytes(unsigned bytes)
{
alSourcei(source, AL_BYTE_OFFSET, bytes);
ALenum error = alGetError();
}