//========= 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; }