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

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//////////////////////////////////////////////////////////////////////////////
//                                                                          //
//  plDSoundBuffer - Simple wrapper class for a DirectSound buffer.         //
//                   Allows us to simplify all the work done behind the     //
//                   scenes in plWin32BufferThread.                         //
//                                                                          //
//////////////////////////////////////////////////////////////////////////////

#include "HeadSpin.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_t plDSoundBuffer::fNumBuffers = 0;
plProfile_CreateCounterNoReset( "Playing", "Sound", SoundPlaying );
plProfile_CreateCounterNoReset( "Allocated", "Sound", NumAllocated );


//// Constructor/Destructor //////////////////////////////////////////////////

plDSoundBuffer::plDSoundBuffer( uint32_t size, plWAVHeader &bufferDesc, bool enable3D, bool isLooping, bool 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_t size, plWAVHeader &bufferDesc, bool enable3D, bool tryStatic )
{
    // Create a DSound buffer description
    fBufferDesc = 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, 0);
    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, 0);
        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, 0);
    SetScalarVolume(0);
    
    
    alSourcef(source, AL_ROLLOFF_FACTOR, 0.3048);
    error = 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;  

        memcpy(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, 0);
    SetScalarVolume(0);
    
    alSourcef(source, AL_ROLLOFF_FACTOR, 0.3048);
    error = 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();
    bool 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);
    error = alGetError();
    if( error != AL_NO_ERROR )
    {
        return false;
    }
    alSourcei(source, AL_BUFFER, 0);
    error = 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( bool 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 );

}

//// Pause ///////////////////////////////////////////////////////////////////

void plDSoundBuffer::Pause()
{
    if (!source)
        return;
    alSourcePause(source);
    alGetError();
}

//// 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 ///////////////////////////////////////////////////////////////

bool    plDSoundBuffer::IsPlaying( void )
{
    ALint state = AL_STOPPED;
    alGetSourcei(source, AL_SOURCE_STATE, &state);
    alGetError();
    return state == AL_PLAYING;
}

//// IsEAXAccelerated ////////////////////////////////////////////////////////

bool    plDSoundBuffer::IsEAXAccelerated( void ) const
{
    return fEAXSource.IsValid();
}

//// bytePosToMSecs //////////////////////////////////////////////////////////

uint32_t  plDSoundBuffer::bytePosToMSecs( uint32_t bytePos ) const
{
    return (uint32_t)(bytePos * 1000 / (float)fBufferDesc->fAvgBytesPerSec);
}

//// GetBufferBytePos ////////////////////////////////////////////////////////

uint32_t  plDSoundBuffer::GetBufferBytePos( float timeInSecs ) const
{
    hsAssert( fBufferDesc != nil, "Nil buffer description when calling GetBufferBytePos()" );

    uint32_t  uint8_t = (uint32_t)( timeInSecs * (float)fBufferDesc->fNumSamplesPerSec );
    uint8_t *= fBufferDesc->fBlockAlign;

    return uint8_t;
}

//// GetLengthInBytes ////////////////////////////////////////////////////////

uint32_t  plDSoundBuffer::GetLengthInBytes( void ) const
{
    return fBufferSize;
}

//// SetEAXSettings //////////////////////////////////////////////////////////

void    plDSoundBuffer::SetEAXSettings(  plEAXSourceSettings *settings, bool force )
{
    fEAXSource.SetFrom( settings, source, force );
}

//// GetBlockAlign ///////////////////////////////////////////////////////////

uint8_t   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( float 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();
}