Modified source engine (2017) developed by valve and leaked in 2020. Not for commercial purporses
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//========= Copyright Valve Corporation, All rights reserved. ============//
//
// Purpose:
//
//=====================================================================================//
#include "audio_pch.h"
#include "fmtstr.h"
#include "sysexternal.h"
// memdbgon must be the last include file in a .cpp file!!!
#include "tier0/memdbgon.h"
extern bool FUseHighQualityPitch( channel_t *pChannel );
//-----------------------------------------------------------------------------
// These mixers provide an abstraction layer between the audio device and
// mixing/decoding code. They allow data to be decoded and mixed using
// optimized, format sensitive code by calling back into the device that
// controls them.
//-----------------------------------------------------------------------------
//-----------------------------------------------------------------------------
// Purpose: maps mixing to 8-bit mono mixer
//-----------------------------------------------------------------------------
class CAudioMixerWave8Mono : public CAudioMixerWave
{
public:
CAudioMixerWave8Mono( IWaveData *data ) : CAudioMixerWave( data ) {}
virtual int GetMixSampleSize() { return CalcSampleSize(8, 1); }
virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress )
{
pDevice->Mix8Mono( pChannel, (char *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress );
}
};
//-----------------------------------------------------------------------------
// Purpose: maps mixing to 8-bit stereo mixer
//-----------------------------------------------------------------------------
class CAudioMixerWave8Stereo : public CAudioMixerWave
{
public:
CAudioMixerWave8Stereo( IWaveData *data ) : CAudioMixerWave( data ) {}
virtual int GetMixSampleSize( ) { return CalcSampleSize(8, 2); }
virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress )
{
pDevice->Mix8Stereo( pChannel, (char *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress );
}
};
//-----------------------------------------------------------------------------
// Purpose: maps mixing to 16-bit mono mixer
//-----------------------------------------------------------------------------
class CAudioMixerWave16Mono : public CAudioMixerWave
{
public:
CAudioMixerWave16Mono( IWaveData *data ) : CAudioMixerWave( data ) {}
virtual int GetMixSampleSize() { return CalcSampleSize(16, 1); }
virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress )
{
pDevice->Mix16Mono( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress );
}
};
//-----------------------------------------------------------------------------
// Purpose: maps mixing to 16-bit stereo mixer
//-----------------------------------------------------------------------------
class CAudioMixerWave16Stereo : public CAudioMixerWave
{
public:
CAudioMixerWave16Stereo( IWaveData *data ) : CAudioMixerWave( data ) {}
virtual int GetMixSampleSize() { return CalcSampleSize(16, 2); }
virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress )
{
pDevice->Mix16Stereo( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress );
}
};
//-----------------------------------------------------------------------------
// Purpose: Create an appropriate mixer type given the data format
// Input : *data - data access abstraction
// format - pcm or adpcm (1 or 2 -- RIFF format)
// channels - number of audio channels (1 = mono, 2 = stereo)
// bits - bits per sample
// Output : CAudioMixer * abstract mixer type that maps mixing to appropriate code
//-----------------------------------------------------------------------------
CAudioMixer *CreateWaveMixer( IWaveData *data, int format, int nChannels, int bits, int initialStreamPosition )
{
CAudioMixer *pMixer = NULL;
if ( format == WAVE_FORMAT_PCM )
{
if ( nChannels > 1 )
{
if ( bits == 8 )
pMixer = new CAudioMixerWave8Stereo( data );
else
pMixer = new CAudioMixerWave16Stereo( data );
}
else
{
if ( bits == 8 )
pMixer = new CAudioMixerWave8Mono( data );
else
pMixer = new CAudioMixerWave16Mono( data );
}
}
else if ( format == WAVE_FORMAT_ADPCM )
{
return CreateADPCMMixer( data );
}
#if defined( _X360 )
else if ( format == WAVE_FORMAT_XMA )
{
return CreateXMAMixer( data, initialStreamPosition );
}
#endif
else
{
// unsupported format or wav file missing!!!
return NULL;
}
if ( pMixer )
{
Assert( CalcSampleSize(bits, nChannels ) == pMixer->GetMixSampleSize() );
}
else
{
Assert( 0 );
}
return pMixer;
}
//-----------------------------------------------------------------------------
// Purpose: Init the base WAVE mixer.
// Input : *data - data access object
//-----------------------------------------------------------------------------
CAudioMixerWave::CAudioMixerWave( IWaveData *data ) : m_pData(data)
{
CAudioSource *pSource = GetSource();
if ( pSource )
{
pSource->ReferenceAdd( this );
}
m_fsample_index = 0;
m_sample_max_loaded = 0;
m_sample_loaded_index = -1;
m_finished = false;
m_forcedEndSample = 0;
m_delaySamples = 0;
}
//-----------------------------------------------------------------------------
// Purpose: Frees the data access object (we own it after construction)
//-----------------------------------------------------------------------------
CAudioMixerWave::~CAudioMixerWave( void )
{
CAudioSource *pSource = GetSource();
if ( pSource )
{
pSource->ReferenceRemove( this );
}
delete m_pData;
}
bool CAudioMixerWave::IsReadyToMix()
{
return m_pData->IsReadyToMix();
}
//-----------------------------------------------------------------------------
// Purpose: Decode and read the data
// by default we just pass the request on to the data access object
// other mixers may need to buffer or decode the data for some reason
//
// Input : **pData - dest pointer
// sampleCount - number of samples needed
// Output : number of samples available in this batch
//-----------------------------------------------------------------------------
int CAudioMixerWave::GetOutputData( void **pData, int sampleCount, char copyBuf[AUDIOSOURCE_COPYBUF_SIZE] )
{
int samples_loaded;
// clear this out in case the underlying code leaves it unmodified
*pData = NULL;
samples_loaded = m_pData->ReadSourceData( pData, m_sample_max_loaded, sampleCount, copyBuf );
// keep track of total samples loaded
m_sample_max_loaded += samples_loaded;
// keep track of index of last sample loaded
m_sample_loaded_index += samples_loaded;
return samples_loaded;
}
//-----------------------------------------------------------------------------
// Purpose: calls through the wavedata to get the audio source
// Output : CAudioSource
//-----------------------------------------------------------------------------
CAudioSource *CAudioMixerWave::GetSource( void )
{
if ( m_pData )
return &m_pData->Source();
return NULL;
}
//-----------------------------------------------------------------------------
// Purpose: Gets the current sample location in playback (index of next sample
// to be loaded).
// Output : int (samples from start of wave)
//-----------------------------------------------------------------------------
int CAudioMixerWave::GetSamplePosition( void )
{
return m_sample_max_loaded;
}
//-----------------------------------------------------------------------------
// Purpose:
// Input : delaySamples -
//-----------------------------------------------------------------------------
void CAudioMixerWave::SetStartupDelaySamples( int delaySamples )
{
m_delaySamples = delaySamples;
}
// Move the current position to newPosition
void CAudioMixerWave::SetSampleStart( int newPosition )
{
CAudioSource *pSource = GetSource();
if ( pSource )
newPosition = pSource->ZeroCrossingAfter( newPosition );
m_fsample_index = newPosition;
// index of last sample loaded - set to sample at new position
m_sample_loaded_index = newPosition;
m_sample_max_loaded = m_sample_loaded_index + 1;
}
// End playback at newEndPosition
void CAudioMixerWave::SetSampleEnd( int newEndPosition )
{
// forced end of zero means play the whole sample
if ( !newEndPosition )
newEndPosition = 1;
CAudioSource *pSource = GetSource();
if ( pSource )
newEndPosition = pSource->ZeroCrossingBefore( newEndPosition );
// past current position? limit.
if ( newEndPosition < m_fsample_index )
newEndPosition = m_fsample_index;
m_forcedEndSample = newEndPosition;
}
//-----------------------------------------------------------------------------
// Purpose: Skip source data (read but don't mix). The mixer must provide the
// full amount of samples or have silence in its output stream.
//-----------------------------------------------------------------------------
int CAudioMixerWave::SkipSamples( channel_t *pChannel, int sampleCount, int outputRate, int outputOffset )
{
float flTempPitch = pChannel->pitch;
pChannel->pitch = 1.0f;
int nRetVal = MixDataToDevice_( NULL, pChannel, sampleCount, outputRate, outputOffset, true );
pChannel->pitch = flTempPitch;
return nRetVal;
}
// wrapper routine to append without overflowing the temp buffer
static uint AppendToBuffer( char *pBuffer, const char *pSampleData, size_t nBytes, const char *pBufferEnd )
{
#if defined(_WIN32) && !defined(_X360)
// FIXME: Some clients are crashing here. Let's try to detect why.
if ( nBytes > 0 && ( (size_t)pBuffer <= 0xFFF || (size_t)pSampleData <= 0xFFF ) )
{
Warning( "AppendToBuffer received potentially bad values (%p, %p, %u, %p)\n", pBuffer, pSampleData, (int)nBytes, pBufferEnd );
}
#endif
if ( pBufferEnd > pBuffer )
{
size_t nAvail = pBufferEnd - pBuffer;
size_t nCopy = MIN( nBytes, nAvail );
Q_memcpy( pBuffer, pSampleData, nCopy );
return nCopy;
}
else
{
return 0;
}
}
// Load a static copy buffer (g_temppaintbuffer) with the requested number of samples,
// with the first sample(s) in the buffer always set up as the last sample(s) of the previous load.
// Return a pointer to the head of the copy buffer.
// This ensures that interpolating pitch shifters always have the previous sample to reference.
// pChannel: sound's channel data
// sample_load_request: number of samples to load from source data
// pSamplesLoaded: returns the actual number of samples loaded (should always = sample_load_request)
// copyBuf: req'd by GetOutputData, used by some Mixers
// Returns: NULL ptr to data if no samples available, otherwise always fills remainder of copy buffer with
// 0 to pad remainder.
// NOTE: DO NOT MODIFY THIS ROUTINE (KELLYB)
char *CAudioMixerWave::LoadMixBuffer( channel_t *pChannel, int sample_load_request, int *pSamplesLoaded, char copyBuf[AUDIOSOURCE_COPYBUF_SIZE] )
{
int samples_loaded;
char *pSample = NULL;
char *pData = NULL;
int cCopySamps = 0;
// save index of last sample loaded (updated in GetOutputData)
int sample_loaded_index = m_sample_loaded_index;
// get data from source (copyBuf is expected to be available for use)
samples_loaded = GetOutputData( (void **)&pData, sample_load_request, copyBuf );
if ( !samples_loaded && sample_load_request )
{
// none available, bail out
// 360 might not be able to get samples due to latency of loop seek
// could also be the valid EOF for non-loops (caller keeps polling for data, until no more)
AssertOnce( IsX360() || !m_pData->Source().IsLooped() );
*pSamplesLoaded = 0;
return NULL;
}
int samplesize = GetMixSampleSize();
const int nTempCopyBufferSize = ( TEMP_COPY_BUFFER_SIZE * sizeof( portable_samplepair_t ) );
char *pCopy = (char *)g_temppaintbuffer;
const char *pCopyBufferEnd = pCopy + nTempCopyBufferSize;
if ( IsX360() || IsDebug() )
{
// for safety, 360 always validates sample request, due to new xma audio code and possible logic flaws
// PC can expect number of requested samples to be within tolerances due to exisiting aged code
// otherwise buffer overruns cause hard to track random crashes
if ( ( ( sample_load_request + 1 ) * samplesize ) > nTempCopyBufferSize )
{
// make sure requested samples will fit in temp buffer.
// if this assert fails, then pitch is too high (ie: > 2.0) or the sample counters have diverged.
// NOTE: to prevent this, pitch should always be capped in MixDataToDevice (but isn't nor are the sample counters).
DevWarning( "LoadMixBuffer: sample load request %d exceeds buffer sizes\n", sample_load_request );
Assert( 0 );
*pSamplesLoaded = 0;
return NULL;
}
}
// copy all samples from pData to copy buffer, set 0th sample to saved previous sample - this ensures
// interpolation pitch shift routines always have a previous sample to reference.
// copy previous sample(s) to head of copy buffer pCopy
// In some cases, we'll need the previous 2 samples. This occurs when
// Rate < 1.0 - in example below, sample 4.86 - 6.48 requires samples 4-7 (previous samples saved are 4 & 5)
/*
Example:
rate = 0.81, sampleCount = 3 (ie: # of samples to return )
_____load 3______ ____load 3_______ __load 2__
0 1 2 3 4 5 6 7 sample_index (whole samples)
^ ^ ^ ^ ^ ^ ^ ^ ^
| | | | | | | | |
0 0.81 1.68 2.43 3.24 4.05 4.86 5.67 6.48 m_fsample_index (rate*sample)
_______________ ________________ ________________
^ ^ ^ ^
| | | |
m_sample_loaded_index | | m_sample_loaded_index
| |
m_fsample_index---- ----m_fsample_index
[return 3 samp] [return 3 samp] [return 3 samp]
*/
pSample = &(pChannel->sample_prev[0]);
// determine how many saved samples we need to copy to head of copy buffer (0,1 or 2)
// so that pitch interpolation will correctly reference samples.
// NOTE: pitch interpolators always reference the sample before and after the indexed sample.
// cCopySamps = sample_max_loaded - floor(m_fsample_index);
if ( sample_loaded_index < 0 || (floor(m_fsample_index) > sample_loaded_index))
{
// no samples previously loaded, or
// next sample index is entirely within the next block of samples to be loaded,
// so we won't need any samples from the previous block. (can occur when rate > 2.0)
cCopySamps = 0;
}
else if ( m_fsample_index < sample_loaded_index )
{
// next sample index is entirely within the previous block of samples loaded,
// so we'll need the last 2 samples loaded. (can occur when rate < 1.0)
Assert ( ceil(m_fsample_index + 0.00000001) == sample_loaded_index );
cCopySamps = 2;
}
else
{
// next sample index is between the next block and the previously loaded block,
// so we'll need the last sample loaded. (can occur when 1.0 < rate < 2.0)
Assert( floor(m_fsample_index) == sample_loaded_index );
cCopySamps = 1;
}
Assert( cCopySamps >= 0 && cCopySamps <= 2 );
// point to the sample(s) we are to copy
if ( cCopySamps )
{
pSample = cCopySamps == 1 ? pSample + samplesize : pSample;
pCopy += AppendToBuffer( pCopy, pSample, samplesize * cCopySamps, pCopyBufferEnd );
}
// copy loaded samples from pData into pCopy
// and update pointer to free space in copy buffer
if ( ( samples_loaded * samplesize ) != 0 && !pData )
{
char const *pWavName = "";
CSfxTable *source = pChannel->sfx;
if ( source )
{
pWavName = source->getname();
}
Warning( "CAudioMixerWave::LoadMixBuffer: '%s' samples_loaded * samplesize = %i but pData == NULL\n", pWavName, ( samples_loaded * samplesize ) );
*pSamplesLoaded = 0;
return NULL;
}
pCopy += AppendToBuffer( pCopy, pData, samples_loaded * samplesize, pCopyBufferEnd );
// if we loaded fewer samples than we wanted to, and we're not
// delaying, load more samples or, if we run out of samples from non-looping source,
// pad copy buffer.
if ( samples_loaded < sample_load_request )
{
// retry loading source data until 0 bytes returned, or we've loaded enough data.
// if we hit 0 bytes, fill remaining space in copy buffer with 0 and exit
int samples_load_extra;
int samples_loaded_retry = -1;
for ( int k = 0; (k < 10000 && samples_loaded_retry && samples_loaded < sample_load_request); k++ )
{
// how many more samples do we need to satisfy load request
samples_load_extra = sample_load_request - samples_loaded;
samples_loaded_retry = GetOutputData( (void**)&pData, samples_load_extra, copyBuf );
// copy loaded samples from pData into pCopy
if ( samples_loaded_retry )
{
if ( ( samples_loaded_retry * samplesize ) != 0 && !pData )
{
Warning( "CAudioMixerWave::LoadMixBuffer: samples_loaded_retry * samplesize = %i but pData == NULL\n", ( samples_loaded_retry * samplesize ) );
*pSamplesLoaded = 0;
return NULL;
}
pCopy += AppendToBuffer( pCopy, pData, samples_loaded_retry * samplesize, pCopyBufferEnd );
samples_loaded += samples_loaded_retry;
}
}
}
// if we still couldn't load the requested samples, fill rest of copy buffer with 0
if ( samples_loaded < sample_load_request )
{
// should always be able to get as many samples as we request from looping sound sources
AssertOnce ( IsX360() || !m_pData->Source().IsLooped() );
// these samples are filled with 0, not loaded.
// non-looping source hit end of data, fill rest of g_temppaintbuffer with 0
int samples_zero_fill = sample_load_request - samples_loaded;
int nAvail = pCopyBufferEnd - pCopy;
int nFill = samples_zero_fill * samplesize;
nFill = MIN( nAvail, nFill );
Q_memset( pCopy, 0, nFill );
pCopy += nFill;
samples_loaded += samples_zero_fill;
}
if ( samples_loaded >= 2 )
{
// always save last 2 samples from copy buffer to channel
// (we'll need 0,1 or 2 samples as start of next buffer for interpolation)
Assert( sizeof( pChannel->sample_prev ) >= samplesize*2 );
pSample = pCopy - samplesize*2;
Q_memcpy( &(pChannel->sample_prev[0]), pSample, samplesize*2 );
}
// this routine must always return as many samples loaded (or zeros) as requested.
Assert( samples_loaded == sample_load_request );
*pSamplesLoaded = samples_loaded;
return (char *)g_temppaintbuffer;
}
// Helper routine to round (rate * samples) down to fixed point precision
double RoundToFixedPoint( double rate, int samples, bool bInterpolated_pitch )
{
fixedint fixp_rate;
int64 d64_newSamps; // need to use double precision int to avoid overflow
double newSamps;
// get rate, in fixed point, determine new samples at rate
if ( bInterpolated_pitch )
fixp_rate = FIX_FLOAT14(rate); // 14 bit iterator
else
fixp_rate = FIX_FLOAT(rate); // 28 bit iterator
// get number of new samples, convert back to float
d64_newSamps = (int64)fixp_rate * (int64)samples;
if ( bInterpolated_pitch )
newSamps = FIX_14TODOUBLE(d64_newSamps);
else
newSamps = FIX_TODOUBLE(d64_newSamps);
return newSamps;
}
extern double MIX_GetMaxRate( double rate, int sampleCount );
// Helper routine for MixDataToDevice:
// Compute number of new samples to load at 'rate' so we can
// output 'sampleCount' samples, from m_fsample_index to fsample_index_end (inclusive)
// rate: sample rate
// sampleCountOut: number of samples calling routine needs to output
// bInterpolated_pitch: true if mixers use interpolating pitch shifters
int CAudioMixerWave::GetSampleLoadRequest( double rate, int sampleCountOut, bool bInterpolated_pitch )
{
double fsample_index_end; // index of last sample we'll need
int sample_index_high; // rounded up last sample index
int sample_load_request; // number of samples to load
// NOTE: we must use fixed point math here, identical to math in mixers, to make sure
// we predict iteration results exactly.
// get floating point sample index of last sample we'll need
fsample_index_end = m_fsample_index + RoundToFixedPoint( rate, sampleCountOut-1, bInterpolated_pitch );
// always round up to ensure we'll have that n+1 sample for interpolation
sample_index_high = (int)( ceil( fsample_index_end ) );
// make sure we always round the floating point index up by at least 1 sample,
// ie: make sure integer sample_index_high is greater than floating point sample index
if ( (double)sample_index_high <= fsample_index_end )
{
sample_index_high++;
}
Assert ( sample_index_high > fsample_index_end );
// attempt to load enough samples so we can reach sample_index_high sample.
sample_load_request = sample_index_high - m_sample_loaded_index;
Assert( sample_index_high >= m_sample_loaded_index );
// NOTE: we can actually return 0 samples to load if rate < 1.0
// and sampleCountOut == 1. In this case, the output sample
// is computed from the previously saved buffer data.
return sample_load_request;
}
int CAudioMixerWave::MixDataToDevice( IAudioDevice *pDevice, channel_t *pChannel, int sampleCount, int outputRate, int outputOffset )
{
return MixDataToDevice_(pDevice, pChannel, sampleCount, outputRate, outputOffset, false );
}
//-----------------------------------------------------------------------------
// Purpose: The device calls this to request data. The mixer must provide the
// full amount of samples or have silence in its output stream.
// Mix channel to all active paintbuffers.
// NOTE: cannot be called consecutively to mix into multiple paintbuffers!
// Input : *pDevice - requesting device
// sampleCount - number of samples at the output rate - should never be more than size of paintbuffer.
// outputRate - sampling rate of the request
// outputOffset - starting offset to mix to in paintbuffer
// bskipallmixing - true if we just want to skip ahead in source data
// Output : Returns true to keep mixing, false to delete this mixer
// NOTE: DO NOT MODIFY THIS ROUTINE (KELLYB)
//-----------------------------------------------------------------------------
int CAudioMixerWave::MixDataToDevice_( IAudioDevice *pDevice, channel_t *pChannel, int sampleCount, int outputRate, int outputOffset, bool bSkipAllMixing )
{
// shouldn't be playing this if finished, but return if we are
if ( m_finished )
return 0;
tmZone( TELEMETRY_LEVEL0, TMZF_NONE, "%s", __FUNCTION__ );
// save this to compute total output
int startingOffset = outputOffset;
double inputRate = (pChannel->pitch * m_pData->Source().SampleRate());
double rate_max = inputRate / outputRate;
// If we are terminating this wave prematurely, then make sure we detect the limit
if ( m_forcedEndSample )
{
// How many total input samples will we need?
int samplesRequired = (int)(sampleCount * rate_max);
// will this hit the end?
if ( m_fsample_index + samplesRequired >= m_forcedEndSample )
{
// yes, mark finished and truncate the sample request
m_finished = true;
sampleCount = (int)( (m_forcedEndSample - m_fsample_index) / rate_max );
}
}
/*
Example:
rate = 1.2, sampleCount = 3 (ie: # of samples to return )
______load 4 samples_____ ________load 4 samples____ ___load 3 samples__
0 1 2 3 4 5 6 7 8 9 10 sample_index (whole samples)
^ ^ ^ ^ ^ ^ ^ ^ ^
| | | | | | | | |
0 1.2 2.4 3.6 4.8 6.0 7.2 8.4 9.6 m_fsample_index (rate*sample)
_______return 3_______ _______return 3_______ _______return 3__________
^ ^
| |
m_sample_loaded_index----- | (after first load 4 samples, this is where pointers are)
m_fsample_index---------
*/
while ( sampleCount > 0 )
{
bool advanceSample = true;
int samples_loaded, outputSampleCount;
char *pData = NULL;
double fsample_index_prev = m_fsample_index; // save so we can modify in LoadMixBuffer
bool bInterpolated_pitch = FUseHighQualityPitch( pChannel );
double rate;
VPROF_( bInterpolated_pitch ? "CAudioMixerWave::MixData innerloop interpolated" : "CAudioMixerWave::MixData innerloop not interpolated", 2, VPROF_BUDGETGROUP_OTHER_SOUND, false, BUDGETFLAG_OTHER );
// process samples in paintbuffer-sized batches
int sampleCountOut = min( sampleCount, PAINTBUFFER_SIZE );
// cap rate so that we never overflow the input copy buffer.
rate = MIX_GetMaxRate( rate_max, sampleCountOut );
if ( m_delaySamples > 0 )
{
// If we are preceding sample playback with a delay,
// just fill data buffer with 0 value samples.
// Because there is no pitch shift applied, outputSampleCount == sampleCountOut.
int num_zero_samples = min( m_delaySamples, sampleCountOut );
// Decrement delay counter
m_delaySamples -= num_zero_samples;
int sampleSize = GetMixSampleSize();
int readBytes = sampleSize * num_zero_samples;
// make sure we don't overflow temp copy buffer (g_temppaintbuffer)
Assert ( (TEMP_COPY_BUFFER_SIZE * sizeof(portable_samplepair_t)) > readBytes );
pData = (char *)g_temppaintbuffer;
// Now copy in some zeroes
memset( pData, 0, readBytes );
// we don't pitch shift these samples, so outputSampleCount == samples_loaded
samples_loaded = num_zero_samples;
outputSampleCount = num_zero_samples;
advanceSample = false;
// the zero samples are at the output rate, so set the input/output ratio to 1.0
rate = 1.0f;
}
else
{
// ask the source for the data...
// temp buffer req'd by some data loaders
char copyBuf[AUDIOSOURCE_COPYBUF_SIZE];
// compute number of new samples to load at 'rate' so we can
// output 'sampleCount' samples, from m_fsample_index to fsample_index_end (inclusive)
int sample_load_request = GetSampleLoadRequest( rate, sampleCountOut, bInterpolated_pitch );
// return pointer to a new copy buffer (g_temppaintbuffer) loaded with sample_load_request samples +
// first sample(s), which are always the last sample(s) from the previous load.
// Always returns sample_load_request samples. Updates m_sample_max_loaded, m_sample_loaded_index.
pData = LoadMixBuffer( pChannel, sample_load_request, &samples_loaded, copyBuf );
// LoadMixBuffer should always return requested samples.
Assert ( !pData || ( samples_loaded == sample_load_request ) );
outputSampleCount = sampleCountOut;
}
// no samples available
if ( !pData )
{
break;
}
// get sample fraction from 0th sample in copy buffer
double sampleFraction = m_fsample_index - floor( m_fsample_index );
// if just skipping samples in source, don't mix, just keep reading
if ( !bSkipAllMixing )
{
// mix this data to all active paintbuffers
// Verify that we won't get a buffer overrun.
Assert( floor( sampleFraction + RoundToFixedPoint(rate, (outputSampleCount-1), bInterpolated_pitch) ) <= samples_loaded );
int saveIndex = MIX_GetCurrentPaintbufferIndex();
for ( int i = 0 ; i < g_paintBuffers.Count(); i++ )
{
if ( g_paintBuffers[i].factive )
{
// mix channel into all active paintbuffers
MIX_SetCurrentPaintbuffer( i );
Mix(
pDevice, // Device.
pChannel, // Channel.
pData, // Input buffer.
outputOffset, // Output position.
FIX_FLOAT( sampleFraction ), // Iterators.
FIX_FLOAT( rate ),
outputSampleCount,
0 );
}
}
MIX_SetCurrentPaintbuffer( saveIndex );
}
if ( advanceSample )
{
// update sample index to point to the next sample to output
// if we're not delaying
// Use fixed point math to make sure we exactly match results of mix
// iterators.
m_fsample_index = fsample_index_prev + RoundToFixedPoint( rate, outputSampleCount, bInterpolated_pitch );
}
outputOffset += outputSampleCount;
sampleCount -= outputSampleCount;
}
// Did we run out of samples? if so, mark finished
if ( sampleCount > 0 )
{
m_finished = true;
}
// total number of samples mixed !!! at the output clock rate !!!
return outputOffset - startingOffset;
}
bool CAudioMixerWave::ShouldContinueMixing( void )
{
return !m_finished;
}
float CAudioMixerWave::ModifyPitch( float pitch )
{
return pitch;
}
float CAudioMixerWave::GetVolumeScale( void )
{
return 1.0f;
}