1

soundsystem/snd_audio_source.cpp

@@ -0,0 +1,58 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//=============================================================================//
+#include <stdio.h>
+#include "soundsystem/snd_audio_source.h"
+#include "soundsystem/isoundsystem.h"
+#include "soundsystem.h"
+
+
+
+extern CAudioSource *Audio_CreateMemoryWave( const char *pName );
+
+//-----------------------------------------------------------------------------
+// Purpose: Simple wrapper to crack naming convention and create the proper wave source
+// Input  : *pName - WAVE filename
+// Output : CAudioSource
+//-----------------------------------------------------------------------------
+CAudioSource *AudioSource_Create( const char *pName )
+{
+	if ( !pName )
+		return NULL;
+
+//	if ( pName[0] == '!' )		// sentence
+		;
+
+	// Names that begin with "*" are streaming.
+	// Skip over the * and create a streamed source
+	if ( pName[0] == '*' )
+	{
+
+		return NULL;
+	}
+
+	// These are loaded into memory directly
+	return Audio_CreateMemoryWave( pName );
+}
+
+CAudioSource::~CAudioSource( void )
+{
+	CAudioMixer *mixer;
+	
+	while ( 1 )
+	{
+		mixer = g_pSoundSystem->FindMixer( this );
+		if ( !mixer )
+			break;
+
+		g_pSoundSystem->StopSound( mixer );
+	}
+}
+
+CAudioSource::CAudioSource( void )
+{
+}

soundsystem/snd_dev_wave.cpp

@@ -0,0 +1,899 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//===========================================================================//
+
+#include "snd_dev_wave.h"
+
+#define WIN32_LEAN_AND_MEAN
+#include <windows.h>
+#pragma warning( disable: 4201 )
+#include <mmsystem.h>
+#pragma warning( default: 4201 )
+#include <stdio.h>
+#include <math.h>
+#include "soundsystem/snd_audio_source.h"
+#include "soundsystem.h"
+#include "soundsystem/snd_device.h"
+#include "tier1/utlvector.h"
+#include "filesystem.h"
+#include "sentence.h"
+
+
+//-----------------------------------------------------------------------------
+// Forward declarations
+//-----------------------------------------------------------------------------
+class CAudioMixer;
+
+
+//-----------------------------------------------------------------------------
+// Important constants
+//-----------------------------------------------------------------------------
+
+// 64K is > 1 second at 16-bit, 22050 Hz
+// 44k: UNDONE - need to double buffers now that we're playing back at 44100?
+#define OUTPUT_CHANNEL_COUNT	2
+#define BYTES_PER_SAMPLE		2
+#define OUTPUT_SAMPLE_RATE		SOUND_DMA_SPEED
+#define OUTPUT_BUFFER_COUNT		64
+#define OUTPUT_BUFFER_MASK		0x3F
+#define OUTPUT_BUFFER_SAMPLE_COUNT	(OUTPUT_BUFFER_SIZE_BYTES / BYTES_PER_SAMPLE)
+#define OUTPUT_BUFFER_SIZE_BYTES	1024
+#define	PAINTBUFFER_SIZE		1024
+#define MAX_CHANNELS			16
+
+
+//-----------------------------------------------------------------------------
+// Implementation of IAudioDevice for WAV files
+//-----------------------------------------------------------------------------
+class CAudioDeviceWave : public IAudioDevice
+{
+public:
+	// Inherited from IAudioDevice
+	virtual bool	Init( void );
+	virtual void	Shutdown( void );
+	virtual const char *DeviceName( void ) const;
+	virtual int		DeviceChannels( void ) const;
+	virtual int		DeviceSampleBits( void ) const;
+	virtual int		DeviceSampleBytes( void ) const;
+	virtual int		DeviceSampleRate( void ) const;
+	virtual int		DeviceSampleCount( void ) const;
+	virtual void	Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward = true );
+	virtual void	Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward = true );
+	virtual void	Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward = true );
+	virtual void	Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward = true );
+	virtual int		PaintBufferSampleCount( void ) const;
+	virtual void	MixBegin( void );
+
+	// mix a buffer up to time (time is absolute)
+	void			Update( float time );
+	void			Flush( void );
+	void			TransferBufferStereo16( short *pOutput, int sampleCount );
+	int				GetOutputPosition( void );
+	float			GetAmountofTimeAhead( void );
+	int				GetNumberofSamplesAhead( void );
+	void			AddSource( CAudioMixer *pSource );
+	void			StopSounds( void );
+	int				FindSourceIndex( CAudioMixer *pSource );
+	CAudioMixer		*GetMixerForSource( CAudioSource *source );
+
+private:
+	class CAudioMixerState
+	{
+	public:
+		CAudioMixer		*mixer;
+	};
+
+	class CAudioBuffer
+	{
+	public:
+		WAVEHDR			*hdr;
+		bool			submitted;
+		int				submit_sample_count;
+
+		CUtlVector< CAudioMixerState > m_Referenced;
+	};
+
+	struct portable_samplepair_t
+	{
+		int left;
+		int	right;
+	};
+
+	void	OpenWaveOut( void );
+	void	CloseWaveOut( void );
+	void	AllocateOutputBuffers();
+	void	FreeOutputBuffers();
+	void*	AllocOutputMemory( int nSize, HGLOBAL &hMemory );
+	void	FreeOutputMemory( HGLOBAL &hMemory );
+
+	bool	ValidWaveOut( void ) const;
+	CAudioBuffer *GetEmptyBuffer( void );
+	void	SilenceBuffer( short *pSamples, int sampleCount );
+
+	void	SetChannel( int channelIndex, CAudioMixer *pSource );
+	void	FreeChannel( int channelIndex );
+
+	void	RemoveMixerChannelReferences( CAudioMixer *mixer );
+	void	AddToReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer );
+	void	RemoveFromReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer );
+	bool	IsSourceReferencedByActiveBuffer( CAudioMixer *mixer );
+	bool	IsSoundInReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer );
+
+	// Compute how many samples we've mixed since most recent buffer submission
+	void	ComputeSampleAheadAmount( void );
+
+	// This is a single allocation for all wave headers (there are OUTPUT_BUFFER_COUNT of them)
+	HGLOBAL			m_hWaveHdr;
+
+	// This is a single allocation for all wave data (there are OUTPUT_BUFFER_COUNT of them)
+	HANDLE			m_hWaveData;
+
+	HWAVEOUT		m_waveOutHandle;
+	float			m_mixTime;
+	float			m_baseTime;
+	int				m_sampleIndex;
+	CAudioBuffer	m_buffers[ OUTPUT_BUFFER_COUNT ];
+	CAudioMixer		*m_sourceList[MAX_CHANNELS];
+	int				m_nEstimatedSamplesAhead;
+
+	portable_samplepair_t		m_paintbuffer[ PAINTBUFFER_SIZE ];
+};
+
+
+//-----------------------------------------------------------------------------
+// Singleton
+//-----------------------------------------------------------------------------
+IAudioDevice *Audio_CreateWaveDevice( void )
+{
+	return new CAudioDeviceWave;
+}
+
+
+//-----------------------------------------------------------------------------
+// Init, shutdown
+//-----------------------------------------------------------------------------
+bool CAudioDeviceWave::Init( void )
+{
+	m_hWaveData = NULL;
+	m_hWaveHdr = NULL;
+	m_waveOutHandle = NULL;
+
+	for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
+	{
+		CAudioBuffer *buffer = &m_buffers[ i ];
+		Assert( buffer );
+		buffer->hdr = NULL;
+		buffer->submitted = false;
+		buffer->submit_sample_count = false;
+	}
+
+	OpenWaveOut();
+
+	m_mixTime = m_baseTime = -1;
+	m_sampleIndex = 0;
+	memset( m_sourceList, 0, sizeof(m_sourceList) );
+
+	m_nEstimatedSamplesAhead = (int)( ( float ) OUTPUT_SAMPLE_RATE / 10.0f );
+
+	return true;
+}
+
+void CAudioDeviceWave::Shutdown( void )
+{
+	CloseWaveOut();
+}
+
+	
+//-----------------------------------------------------------------------------
+// WAV out device
+//-----------------------------------------------------------------------------
+inline bool CAudioDeviceWave::ValidWaveOut( void ) const 
+{ 
+	return m_waveOutHandle != 0; 
+}
+
+
+//-----------------------------------------------------------------------------
+// Opens the windows wave out device
+//-----------------------------------------------------------------------------
+void CAudioDeviceWave::OpenWaveOut( void )
+{
+	WAVEFORMATEX waveFormat;
+	memset( &waveFormat, 0, sizeof(waveFormat) );
+
+	// Select a PCM, 16-bit stereo playback device
+	waveFormat.cbSize = sizeof(waveFormat);
+	waveFormat.wFormatTag = WAVE_FORMAT_PCM;
+	waveFormat.nChannels = DeviceChannels();
+	waveFormat.wBitsPerSample = DeviceSampleBits();
+	waveFormat.nSamplesPerSec = DeviceSampleRate();
+	waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
+	waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign; 
+
+	MMRESULT errorCode = waveOutOpen( &m_waveOutHandle, WAVE_MAPPER, &waveFormat, 0, 0L, CALLBACK_NULL );
+	while ( errorCode != MMSYSERR_NOERROR )
+	{
+		if ( errorCode != MMSYSERR_ALLOCATED )
+		{
+			DWarning( "soundsystem", 1, "waveOutOpen failed\n" );
+			m_waveOutHandle = 0;
+			return;
+		}
+
+		int nRetVal = MessageBox( NULL,
+			"The sound hardware is in use by another app.\n\n"
+			"Select Retry to try to start sound again or Cancel to run with no sound.",
+			"Sound not available",
+			MB_RETRYCANCEL | MB_SETFOREGROUND | MB_ICONEXCLAMATION);
+
+		if ( nRetVal != IDRETRY )
+		{
+			DWarning( "soundsystem", 1, "waveOutOpen failure--hardware already in use\n" );
+			m_waveOutHandle = 0;
+			return;
+		}
+
+		errorCode = waveOutOpen( &m_waveOutHandle, WAVE_MAPPER, &waveFormat, 0, 0L, CALLBACK_NULL );
+	}
+
+	AllocateOutputBuffers();
+}
+
+
+//-----------------------------------------------------------------------------
+// Closes the windows wave out device
+//-----------------------------------------------------------------------------
+void CAudioDeviceWave::CloseWaveOut( void ) 
+{ 
+	if ( ValidWaveOut() )
+	{
+		waveOutReset( m_waveOutHandle );
+		FreeOutputBuffers();
+		waveOutClose( m_waveOutHandle );
+		m_waveOutHandle = NULL; 
+	}
+}
+
+
+//-----------------------------------------------------------------------------
+// Alloc output memory
+//-----------------------------------------------------------------------------
+void* CAudioDeviceWave::AllocOutputMemory( int nSize, HGLOBAL &hMemory )
+{
+	// Output memory for waveform data+hdrs must be 
+	// globally allocated with GMEM_MOVEABLE and GMEM_SHARE flags.
+	hMemory = GlobalAlloc( GMEM_MOVEABLE | GMEM_SHARE, nSize ); 
+	if ( !hMemory ) 
+	{ 
+		DWarning( "soundsystem", 1, "Sound: Out of memory.\n");
+		CloseWaveOut();
+		return NULL;
+	}
+
+	HPSTR lpData = (char *)GlobalLock( hMemory );
+	if ( !lpData )
+	{ 
+		DWarning( "soundsystem", 1, "Sound: Failed to lock.\n");
+		GlobalFree( hMemory );
+		hMemory = NULL;
+		CloseWaveOut();
+		return NULL;
+	} 
+	memset( lpData, 0, nSize );
+	return lpData;
+}
+
+
+//-----------------------------------------------------------------------------
+// Free output memory
+//-----------------------------------------------------------------------------
+void CAudioDeviceWave::FreeOutputMemory( HGLOBAL &hMemory )
+{
+	if ( hMemory )
+	{
+		GlobalUnlock( hMemory ); 
+		GlobalFree( hMemory );
+		hMemory = NULL;
+	}
+}
+
+
+//-----------------------------------------------------------------------------
+// Allocate, free output buffers
+//-----------------------------------------------------------------------------
+void CAudioDeviceWave::AllocateOutputBuffers()
+{
+	// Allocate and lock memory for the waveform data.  
+	int nBufferSize = OUTPUT_BUFFER_SIZE_BYTES * OUTPUT_BUFFER_COUNT;
+	HPSTR lpData = (char *)AllocOutputMemory( nBufferSize, m_hWaveData );
+	if ( !lpData )
+		return;
+
+	// Allocate and lock memory for the waveform header
+	int nHdrSize = sizeof( WAVEHDR ) * OUTPUT_BUFFER_COUNT;
+	LPWAVEHDR lpWaveHdr = (LPWAVEHDR)AllocOutputMemory( nHdrSize, m_hWaveHdr );
+	if ( !lpWaveHdr )
+		return;
+
+	// After allocation, set up and prepare headers.
+	for ( int i=0 ; i < OUTPUT_BUFFER_COUNT; i++ )
+	{
+		LPWAVEHDR lpHdr = lpWaveHdr + i;
+		lpHdr->dwBufferLength = OUTPUT_BUFFER_SIZE_BYTES; 
+		lpHdr->lpData = lpData + (i * OUTPUT_BUFFER_SIZE_BYTES);
+
+		MMRESULT nResult = waveOutPrepareHeader( m_waveOutHandle, lpHdr, sizeof(WAVEHDR) );
+		if ( nResult != MMSYSERR_NOERROR )
+		{
+			DWarning( "soundsystem", 1, "Sound: failed to prepare wave headers\n" );
+			CloseWaveOut();
+			return;
+		}
+
+		m_buffers[i].hdr = lpHdr;
+	}
+}
+
+
+void CAudioDeviceWave::FreeOutputBuffers()
+{
+	// Unprepare headers.
+	for ( int i=0 ; i < OUTPUT_BUFFER_COUNT; i++ )
+	{
+		if ( m_buffers[i].hdr )
+		{
+			waveOutUnprepareHeader( m_waveOutHandle, m_buffers[i].hdr, sizeof(WAVEHDR) );
+			m_buffers[i].hdr = NULL;
+		}
+
+		m_buffers[i].submitted = false;
+		m_buffers[i].submit_sample_count = 0;
+		m_buffers[i].m_Referenced.Purge();
+	}
+
+	FreeOutputMemory( m_hWaveData );
+	FreeOutputMemory( m_hWaveHdr );
+}
+
+	
+//-----------------------------------------------------------------------------
+// Device parameters
+//-----------------------------------------------------------------------------
+const char *CAudioDeviceWave::DeviceName( void ) const			
+{ 
+	return "Windows WAVE"; 
+}
+
+int CAudioDeviceWave::DeviceChannels( void ) const		
+{ 
+	return 2; 
+}
+
+int CAudioDeviceWave::DeviceSampleBits( void ) const	
+{ 
+	return (BYTES_PER_SAMPLE * 8); 
+}
+
+int CAudioDeviceWave::DeviceSampleBytes( void ) const	
+{ 
+	return BYTES_PER_SAMPLE; 
+}
+
+int CAudioDeviceWave::DeviceSampleRate( void ) const		
+{ 
+	return OUTPUT_SAMPLE_RATE; 
+}
+
+int CAudioDeviceWave::DeviceSampleCount( void )	const
+{ 
+	return OUTPUT_BUFFER_SAMPLE_COUNT; 
+}
+
+int CAudioDeviceWave::PaintBufferSampleCount( void ) const
+{
+	return PAINTBUFFER_SIZE;
+}
+
+
+//-----------------------------------------------------------------------------
+// Mixing routines
+//-----------------------------------------------------------------------------
+void CAudioDeviceWave::Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward )
+{
+	int sampleIndex = 0;
+	fixedint sampleFrac = inputOffset;
+
+	int fixup = 0;
+	int fixupstep = 1;
+
+	if ( !forward )
+	{
+		fixup = outCount - 1;
+		fixupstep = -1;
+	}
+
+	for ( int i = 0; i < outCount; i++, fixup += fixupstep )
+	{
+		int dest = max( outputOffset + fixup, 0 );
+
+		m_paintbuffer[ dest ].left += pChannel->leftvol * pData[sampleIndex];
+		m_paintbuffer[ dest ].right += pChannel->rightvol * pData[sampleIndex];
+		sampleFrac += rateScaleFix;
+		sampleIndex += FIX_INTPART(sampleFrac);
+		sampleFrac = FIX_FRACPART(sampleFrac);
+	}
+}
+
+
+void CAudioDeviceWave::Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward )
+{
+	int sampleIndex = 0;
+	fixedint sampleFrac = inputOffset;
+
+	int fixup = 0;
+	int fixupstep = 1;
+
+	if ( !forward )
+	{
+		fixup = outCount - 1;
+		fixupstep = -1;
+	}
+
+	for ( int i = 0; i < outCount; i++, fixup += fixupstep )
+	{
+		int dest = max( outputOffset + fixup, 0 );
+
+		m_paintbuffer[ dest ].left += pChannel->leftvol * pData[sampleIndex];
+		m_paintbuffer[ dest ].right += pChannel->rightvol * pData[sampleIndex+1];
+		sampleFrac += rateScaleFix;
+		sampleIndex += FIX_INTPART(sampleFrac)<<1;
+		sampleFrac = FIX_FRACPART(sampleFrac);
+	}
+}
+
+
+void CAudioDeviceWave::Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward )
+{
+	int sampleIndex = 0;
+	fixedint sampleFrac = inputOffset;
+
+	int fixup = 0;
+	int fixupstep = 1;
+
+	if ( !forward )
+	{
+		fixup = outCount - 1;
+		fixupstep = -1;
+	}
+
+	for ( int i = 0; i < outCount; i++, fixup += fixupstep )
+	{
+		int dest = max( outputOffset + fixup, 0 );
+
+		m_paintbuffer[ dest ].left += (pChannel->leftvol * pData[sampleIndex])>>8;
+		m_paintbuffer[ dest ].right += (pChannel->rightvol * pData[sampleIndex])>>8;
+		sampleFrac += rateScaleFix;
+		sampleIndex += FIX_INTPART(sampleFrac);
+		sampleFrac = FIX_FRACPART(sampleFrac);
+	}
+}
+
+
+void CAudioDeviceWave::Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress, bool forward )
+{
+	int sampleIndex = 0;
+	fixedint sampleFrac = inputOffset;
+
+	int fixup = 0;
+	int fixupstep = 1;
+
+	if ( !forward )
+	{
+		fixup = outCount - 1;
+		fixupstep = -1;
+	}
+
+	for ( int i = 0; i < outCount; i++, fixup += fixupstep )
+	{
+		int dest = max( outputOffset + fixup, 0 );
+
+		m_paintbuffer[ dest ].left += (pChannel->leftvol * pData[sampleIndex])>>8;
+		m_paintbuffer[ dest ].right += (pChannel->rightvol * pData[sampleIndex+1])>>8;
+
+		sampleFrac += rateScaleFix;
+		sampleIndex += FIX_INTPART(sampleFrac)<<1;
+		sampleFrac = FIX_FRACPART(sampleFrac);
+	}
+}
+
+
+void CAudioDeviceWave::MixBegin( void )
+{
+	memset( m_paintbuffer, 0, sizeof(m_paintbuffer) );
+}
+
+void CAudioDeviceWave::TransferBufferStereo16( short *pOutput, int sampleCount )
+{
+	for ( int i = 0; i < sampleCount; i++ )
+	{
+		if ( m_paintbuffer[i].left > 32767 )
+			m_paintbuffer[i].left = 32767;
+		else if ( m_paintbuffer[i].left < -32768 )
+			m_paintbuffer[i].left = -32768;
+
+		if ( m_paintbuffer[i].right > 32767 )
+			m_paintbuffer[i].right = 32767;
+		else if ( m_paintbuffer[i].right < -32768 )
+			m_paintbuffer[i].right = -32768;
+
+		*pOutput++ = (short)m_paintbuffer[i].left;
+		*pOutput++ = (short)m_paintbuffer[i].right;
+	}
+}
+
+void CAudioDeviceWave::RemoveMixerChannelReferences( CAudioMixer *mixer )
+{
+	for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
+	{
+		RemoveFromReferencedList( mixer, &m_buffers[ i ] );
+	}
+}
+
+void CAudioDeviceWave::AddToReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer )
+{
+	// Already in list
+	for ( int i = 0; i < buffer->m_Referenced.Size(); i++ )
+	{
+		if ( buffer->m_Referenced[ i ].mixer == mixer )
+			return;
+	}
+
+	// Just remove it
+	int idx = buffer->m_Referenced.AddToTail();
+
+	CAudioMixerState *state = &buffer->m_Referenced[ idx ];
+	state->mixer = mixer;
+}
+
+void CAudioDeviceWave::RemoveFromReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer )
+{
+	for ( int i = 0; i < buffer->m_Referenced.Size(); i++ )
+	{
+		if ( buffer->m_Referenced[ i ].mixer == mixer )
+		{
+			buffer->m_Referenced.Remove( i );
+			break;
+		}
+	}
+}
+
+bool CAudioDeviceWave::IsSoundInReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer )
+{
+	for ( int i = 0; i < buffer->m_Referenced.Size(); i++ )
+	{
+		if ( buffer->m_Referenced[ i ].mixer == mixer )
+		{
+			return true;
+		}
+	}
+	return false;
+}
+
+bool CAudioDeviceWave::IsSourceReferencedByActiveBuffer( CAudioMixer *mixer )
+{
+	if ( !ValidWaveOut() )
+		return false;
+
+	CAudioBuffer *buffer;
+	for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
+	{
+		buffer = &m_buffers[ i ];
+		if ( !buffer->submitted )
+			continue;
+
+		if ( buffer->hdr->dwFlags & WHDR_DONE )
+			continue;
+
+		// See if it's referenced
+		if ( IsSoundInReferencedList( mixer, buffer ) )
+			return true;
+	}
+
+	return false;
+}
+
+CAudioDeviceWave::CAudioBuffer *CAudioDeviceWave::GetEmptyBuffer( void )
+{
+	CAudioBuffer *pOutput = NULL;
+	if ( ValidWaveOut() )
+	{
+		for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
+		{
+			if ( !(m_buffers[ i ].submitted ) || 
+				m_buffers[i].hdr->dwFlags & WHDR_DONE )
+			{
+				pOutput = &m_buffers[i];
+				pOutput->submitted = true;
+				pOutput->m_Referenced.Purge();
+				break;
+			}
+		}
+	}
+	
+	return pOutput;
+}
+
+void CAudioDeviceWave::SilenceBuffer( short *pSamples, int sampleCount )
+{
+	int i;
+
+	for ( i = 0; i < sampleCount; i++ )
+	{
+		// left
+		*pSamples++ = 0;
+		// right
+		*pSamples++ = 0;
+	}
+}
+
+void CAudioDeviceWave::Flush( void )
+{
+	waveOutReset( m_waveOutHandle );
+}
+
+// mix a buffer up to time (time is absolute)
+void CAudioDeviceWave::Update( float time )
+{
+	if ( !ValidWaveOut() )
+		return;
+
+	// reset the system
+	if ( m_mixTime < 0 || time < m_baseTime )
+	{
+		m_baseTime = time;
+		m_mixTime = 0;
+	}
+
+	// put time in our coordinate frame
+	time -= m_baseTime;
+
+	if ( time > m_mixTime )
+	{
+		CAudioBuffer *pBuffer = GetEmptyBuffer();
+		
+		// no free buffers, mixing is ahead of the playback!
+		if ( !pBuffer || !pBuffer->hdr )
+		{
+			//Con_Printf( "out of buffers\n" );
+			return;
+		}
+
+		// UNDONE: These numbers are constants
+		// calc number of samples (2 channels * 2 bytes per sample)
+		int sampleCount = pBuffer->hdr->dwBufferLength >> 2;
+		m_mixTime += sampleCount * (1.0f / OUTPUT_SAMPLE_RATE);
+
+		short *pSamples = reinterpret_cast<short *>(pBuffer->hdr->lpData);
+		
+		SilenceBuffer( pSamples, sampleCount );
+
+		int tempCount = sampleCount;
+
+		while ( tempCount > 0 )
+		{
+			if ( tempCount > PaintBufferSampleCount() )
+			{
+				sampleCount = PaintBufferSampleCount();
+			}
+			else
+			{
+				sampleCount = tempCount;
+			}
+
+			MixBegin();
+			for ( int i = 0; i < MAX_CHANNELS; i++ )
+			{
+				CAudioMixer *pSource = m_sourceList[i];
+				if ( !pSource )
+					continue;
+
+				int currentsample = pSource->GetSamplePosition();
+				bool forward = pSource->GetDirection();
+
+				if ( pSource->GetActive() )
+				{
+					if ( !pSource->MixDataToDevice( this, pSource->GetChannel(), currentsample, sampleCount, DeviceSampleRate(), forward ) )
+					{
+						// Source becomes inactive when last submitted sample is finally
+						//  submitted.  But it lingers until it's no longer referenced
+						pSource->SetActive( false );
+					}
+					else
+					{
+						AddToReferencedList( pSource, pBuffer );
+					}
+				} 
+				else 
+				{
+					if ( !IsSourceReferencedByActiveBuffer( pSource ) )
+					{
+						if ( !pSource->GetAutoDelete() )
+						{
+							FreeChannel( i );
+						}
+					}
+					else
+					{
+						pSource->IncrementSamples( pSource->GetChannel(), currentsample, sampleCount, DeviceSampleRate(), forward );
+					}
+				}
+
+			}
+
+			TransferBufferStereo16( pSamples, sampleCount );
+
+			m_sampleIndex += sampleCount;
+			tempCount -= sampleCount;
+			pSamples += sampleCount * 2;
+		}
+		// if the buffers aren't aligned on sample boundaries, this will hard-lock the machine!
+
+		pBuffer->submit_sample_count = GetOutputPosition();
+
+		waveOutWrite( m_waveOutHandle, pBuffer->hdr, sizeof(*(pBuffer->hdr)) );
+	}
+}
+
+/*
+int CAudioDeviceWave::GetNumberofSamplesAhead( void )
+{
+	ComputeSampleAheadAmount();
+	return m_nEstimatedSamplesAhead;
+}
+
+float CAudioDeviceWave::GetAmountofTimeAhead( void )
+{
+	ComputeSampleAheadAmount();
+	return ( (float)m_nEstimatedSamplesAhead / (float)OUTPUT_SAMPLE_RATE );
+}
+
+// Find the most recent submitted sample that isn't flagged as whdr_done
+void CAudioDeviceWave::ComputeSampleAheadAmount( void )
+{
+	m_nEstimatedSamplesAhead = 0;
+
+	int newest_sample_index = -1;
+	int newest_sample_count = 0;
+
+	CAudioBuffer *buffer;
+
+	if ( ValidDevice() )
+	{
+
+		for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
+		{
+			buffer = &m_buffers[ i ];
+			if ( !buffer->submitted )
+				continue;
+
+			if ( buffer->hdr->dwFlags & WHDR_DONE )
+				continue;
+
+			if ( buffer->submit_sample_count > newest_sample_count )
+			{
+				newest_sample_index = i;
+				newest_sample_count = buffer->submit_sample_count;
+			}
+		}
+	}
+
+	if ( newest_sample_index == -1 )
+		return;
+
+
+	buffer = &m_buffers[ newest_sample_index ];
+	int currentPos = GetOutputPosition() ;
+	m_nEstimatedSamplesAhead = currentPos - buffer->submit_sample_count;
+}
+*/
+
+int CAudioDeviceWave::FindSourceIndex( CAudioMixer *pSource )
+{
+	for ( int i = 0; i < MAX_CHANNELS; i++ )
+	{
+		if ( pSource == m_sourceList[i] )
+		{
+			return i;
+		}
+	}
+	return -1;
+}
+
+CAudioMixer *CAudioDeviceWave::GetMixerForSource( CAudioSource *source )
+{
+	for ( int i = 0; i < MAX_CHANNELS; i++ )
+	{
+		if ( !m_sourceList[i] )
+			continue;
+
+		if ( source == m_sourceList[i]->GetSource() )
+		{
+			return m_sourceList[i];
+		}
+	}
+	return NULL;
+}
+
+void CAudioDeviceWave::AddSource( CAudioMixer *pSource )
+{
+	int slot = 0;
+	for ( int i = 0; i < MAX_CHANNELS; i++ )
+	{
+		if ( !m_sourceList[i] )
+		{
+			slot = i;
+			break;
+		}
+	}
+
+	if ( m_sourceList[slot] )
+	{
+		FreeChannel( slot );
+	}
+	SetChannel( slot, pSource );
+
+	pSource->SetActive( true );
+}
+
+
+void CAudioDeviceWave::StopSounds( void )
+{
+	for ( int i = 0; i < MAX_CHANNELS; i++ )
+	{
+		if ( m_sourceList[i] )
+		{
+			FreeChannel( i );
+		}
+	}
+}
+
+
+void CAudioDeviceWave::SetChannel( int channelIndex, CAudioMixer *pSource )
+{
+	if ( channelIndex < 0 || channelIndex >= MAX_CHANNELS )
+		return;
+
+	m_sourceList[channelIndex] = pSource;
+}
+
+void CAudioDeviceWave::FreeChannel( int channelIndex )
+{
+	if ( channelIndex < 0 || channelIndex >= MAX_CHANNELS )
+		return;
+
+	if ( m_sourceList[channelIndex] )
+	{
+		RemoveMixerChannelReferences( m_sourceList[channelIndex] );
+
+		delete m_sourceList[channelIndex];
+		m_sourceList[channelIndex] = NULL;
+	}
+}
+
+int CAudioDeviceWave::GetOutputPosition( void )
+{
+	if ( !m_waveOutHandle )
+		return 0;
+
+	MMTIME mmtime;
+	mmtime.wType = TIME_SAMPLES;
+	waveOutGetPosition( m_waveOutHandle, &mmtime, sizeof( MMTIME ) );
+
+	// Convert time to sample count
+	return ( mmtime.u.sample );
+}
+

soundsystem/snd_dev_wave.h

@@ -0,0 +1,27 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//=============================================================================//
+
+#ifndef SND_DEV_WAVE_H
+#define SND_DEV_WAVE_H
+#ifdef _WIN32
+#pragma once
+#endif
+
+
+//-----------------------------------------------------------------------------
+// Forward declarations
+//-----------------------------------------------------------------------------
+class IAudioDevice;
+
+
+//-----------------------------------------------------------------------------
+// Creates a device that mixes WAVs using windows
+//-----------------------------------------------------------------------------
+IAudioDevice *Audio_CreateWaveDevice( void );
+
+
+#endif // SND_DEV_WAVE_H

soundsystem/snd_io.cpp

@@ -0,0 +1,93 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//===========================================================================//
+
+#include "soundsystem.h"
+#include "tier2/riff.h"
+#include "filesystem.h"
+#include "tier1/strtools.h"
+
+// memdbgon must be the last include file in a .cpp file!!!
+#include "tier0/memdbgon.h"
+
+//-----------------------------------------------------------------------------
+// Purpose: Implements Audio IO on the engine's COMMON filesystem
+//-----------------------------------------------------------------------------
+class COM_IOReadBinary : public IFileReadBinary
+{
+public:
+	int open( const char *pFileName );
+	int read( void *pOutput, int size, int file );
+	void seek( int file, int pos );
+	unsigned int tell( int file );
+	unsigned int size( int file );
+	void close( int file );
+};
+
+
+// prepend sound/ to the filename -- all sounds are loaded from the sound/ directory
+int COM_IOReadBinary::open( const char *pFileName )
+{
+	char namebuffer[512];
+	FileHandle_t hFile;
+
+	Q_strncpy(namebuffer, "sound", sizeof( namebuffer ) );
+
+	//HACK HACK HACK  the server is sending back sound names with slashes in front... 
+	if (pFileName[0]!='/')
+	{
+		Q_strncat(namebuffer,"/", sizeof( namebuffer ), COPY_ALL_CHARACTERS );
+	}
+
+	Q_strncat( namebuffer, pFileName, sizeof( namebuffer ), COPY_ALL_CHARACTERS );
+
+	hFile = g_pFullFileSystem->Open( namebuffer, "rb", "GAME" );
+
+	return (int)hFile;
+}
+
+int COM_IOReadBinary::read( void *pOutput, int size, int file )
+{
+	if ( !file )
+		return 0;
+
+	return g_pFullFileSystem->Read( pOutput, size, (FileHandle_t)file );
+}
+
+void COM_IOReadBinary::seek( int file, int pos )
+{
+	if ( !file )
+		return;
+
+	g_pFullFileSystem->Seek( (FileHandle_t)file, pos, FILESYSTEM_SEEK_HEAD );
+}
+
+unsigned int COM_IOReadBinary::tell( int file )
+{
+	if ( !file )
+		return 0;
+	return g_pFullFileSystem->Tell( (FileHandle_t)file );
+}
+
+unsigned int COM_IOReadBinary::size( int file )
+{
+	if (!file)
+		return 0;
+	return g_pFullFileSystem->Size( (FileHandle_t)file );
+}
+
+void COM_IOReadBinary::close( int file )
+{
+	if (!file)
+		return;
+
+	g_pFullFileSystem->Close( (FileHandle_t)file );
+}
+
+static COM_IOReadBinary io;
+IFileReadBinary *g_pSndIO = &io;
+

soundsystem/snd_wave_mixer.cpp

@@ -0,0 +1,527 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $Workfile:     $
+// $Date:         $
+// $NoKeywords: $
+//===========================================================================//
+#include <stdio.h>
+
+#include "snd_dev_wave.h"
+#include "snd_wave_source.h"
+#include "soundsystem/snd_audio_source.h"
+#include "snd_wave_mixer_private.h"
+#include "snd_wave_mixer_adpcm.h"
+#include "tier2/riff.h"
+
+//-----------------------------------------------------------------------------
+// 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( CWaveData *data ) : CAudioMixerWave( data ) {}
+	virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress, bool forward = true )
+	{
+		pDevice->Mix8Mono( pChannel, (char *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
+	}
+};
+
+//-----------------------------------------------------------------------------
+// Purpose: maps mixing to 8-bit stereo mixer
+//-----------------------------------------------------------------------------
+class CAudioMixerWave8Stereo : public CAudioMixerWave
+{
+public:
+	CAudioMixerWave8Stereo( CWaveData *data ) : CAudioMixerWave( data ) {}
+	virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress, bool forward = true )
+	{
+		pDevice->Mix8Stereo( pChannel, (char *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
+	}
+};
+
+//-----------------------------------------------------------------------------
+// Purpose: maps mixing to 16-bit mono mixer
+//-----------------------------------------------------------------------------
+class CAudioMixerWave16Mono : public CAudioMixerWave
+{
+public:
+	CAudioMixerWave16Mono( CWaveData *data ) : CAudioMixerWave( data ) {}
+	virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress, bool forward = true )
+	{
+		pDevice->Mix16Mono( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
+	}
+};
+
+
+//-----------------------------------------------------------------------------
+// Purpose: maps mixing to 16-bit stereo mixer
+//-----------------------------------------------------------------------------
+class CAudioMixerWave16Stereo : public CAudioMixerWave
+{
+public:
+	CAudioMixerWave16Stereo( CWaveData *data ) : CAudioMixerWave( data ) {}
+	virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress, bool forward = true )
+	{
+		pDevice->Mix16Stereo( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
+	}
+};
+
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Create an approprite 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( CWaveData *data, int format, int channels, int bits )
+{
+	if ( format == WAVE_FORMAT_PCM )
+	{
+		if ( channels > 1 )
+		{
+			if ( bits == 8 )
+				return new CAudioMixerWave8Stereo( data );
+			else
+				return new CAudioMixerWave16Stereo( data );
+		}
+		else
+		{
+			if ( bits == 8 )
+				return new CAudioMixerWave8Mono( data );
+			else
+				return new CAudioMixerWave16Mono( data );
+		}
+	}
+	else if ( format == WAVE_FORMAT_ADPCM )
+	{
+		return CreateADPCMMixer( data );
+	}
+	return NULL;
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Init the base WAVE mixer.
+// Input  : *data - data access object
+//-----------------------------------------------------------------------------
+CAudioMixerWave::CAudioMixerWave( CWaveData *data ) : m_pData(data), m_pChannel(NULL)
+{
+	m_loop = 0;
+	m_sample = 0;
+	m_absoluteSample = 0;
+	m_scubSample = -1;
+	m_fracOffset = 0;
+	m_bActive = false;
+	m_nModelIndex = -1;
+	m_bForward = true;
+	m_bAutoDelete = true;
+	m_pChannel = new channel_t;
+	m_pChannel->leftvol   = 127;
+	m_pChannel->rightvol  = 127;
+	m_pChannel->pitch		= 1.0;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Frees the data access object (we own it after construction)
+//-----------------------------------------------------------------------------
+CAudioMixerWave::~CAudioMixerWave( void )
+{
+	delete m_pData;
+	delete m_pChannel;
+}
+
+
+//-----------------------------------------------------------------------------
+// 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 samplePosition, int sampleCount, bool forward /*= true*/ )
+{
+	if ( samplePosition != m_sample )
+	{
+		// Seek
+		m_sample = samplePosition;
+		m_absoluteSample = samplePosition;
+	}
+
+	return m_pData->ReadSourceData( pData, m_sample, sampleCount, forward );
+}
+
+
+//-----------------------------------------------------------------------------
+// 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
+// Output : int (samples from start of wave)
+//-----------------------------------------------------------------------------
+int CAudioMixerWave::GetSamplePosition( void )
+{
+	return m_sample;
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Gets the current sample location in playback
+// Output : int (samples from start of wave)
+//-----------------------------------------------------------------------------
+int CAudioMixerWave::GetScubPosition( void )
+{
+	if (m_scubSample != -1)
+	{
+		return m_scubSample;
+	}
+	return m_sample;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : position - 
+//-----------------------------------------------------------------------------
+bool CAudioMixerWave::SetSamplePosition( int position, bool scrubbing )
+{
+	position = max( 0, position );
+
+	m_sample = position;
+	m_absoluteSample = position;
+	m_startpos = m_sample;
+	if (scrubbing)
+	{
+		m_scubSample = position;
+	}
+	else
+	{
+		m_scubSample = -1;
+	}
+		
+	return true;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : position - 
+//-----------------------------------------------------------------------------
+void CAudioMixerWave::SetLoopPosition( int position )
+{
+	m_loop = position;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Output : int
+//-----------------------------------------------------------------------------
+int CAudioMixerWave::GetStartPosition( void )
+{
+	return m_startpos;
+}
+
+bool CAudioMixerWave::GetActive( void )
+{
+	return m_bActive;
+}
+
+void CAudioMixerWave::SetActive( bool active )
+{
+	m_bActive = active;
+}
+
+void CAudioMixerWave::SetModelIndex( int index )
+{
+	m_nModelIndex = index;
+}
+
+int CAudioMixerWave::GetModelIndex( void ) const
+{
+	return m_nModelIndex;
+}
+
+void CAudioMixerWave::SetDirection( bool forward )
+{
+	m_bForward = forward;
+}
+
+bool CAudioMixerWave::GetDirection( void ) const
+{
+	return m_bForward;
+}
+
+void CAudioMixerWave::SetAutoDelete( bool autodelete )
+{
+	m_bAutoDelete = autodelete;
+}
+
+bool CAudioMixerWave::GetAutoDelete( void ) const
+{
+	return m_bAutoDelete;
+}
+
+void CAudioMixerWave::SetVolume( float volume )
+{
+	int ivolume = (int)( clamp( volume, 0.0f, 1.0f ) * 127.0f );
+
+	m_pChannel->leftvol   = ivolume;
+	m_pChannel->rightvol  = ivolume;
+}
+
+channel_t *CAudioMixerWave::GetChannel()
+{
+	Assert( m_pChannel );
+	return m_pChannel;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : *pChannel - 
+//			sampleCount - 
+//			outputRate - 
+//-----------------------------------------------------------------------------
+void CAudioMixerWave::IncrementSamples( channel_t *pChannel, int startSample, int sampleCount,int outputRate, bool forward /*= true*/ )
+{
+	int inputSampleRate = (int)(pChannel->pitch * m_pData->Source().SampleRate());
+	float rate = (float)inputSampleRate / outputRate;
+
+	int startpos = startSample;
+
+	if ( !forward )
+	{
+		int requestedstart = startSample - (int)( sampleCount * rate );
+		if ( requestedstart < 0 )
+			return;
+
+		startpos = max( 0, requestedstart );
+		SetSamplePosition( startpos );
+	}
+
+	while ( sampleCount > 0 )
+	{
+		int inputSampleCount;
+		int outputSampleCount = sampleCount;
+		
+		if ( outputRate != inputSampleRate )
+		{
+			inputSampleCount = (int)(sampleCount * rate);
+		}
+		else
+		{
+			inputSampleCount = sampleCount;
+		}
+		
+		sampleCount -= outputSampleCount;
+		if ( forward )
+		{
+			m_sample += inputSampleCount;
+			m_absoluteSample += inputSampleCount;
+		}
+	}
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: The device calls this to request data.  The mixer must provide the
+//			full amount of samples or have silence in its output stream.
+// Input  : *pDevice - requesting device
+//			sampleCount - number of samples at the output rate
+//			outputRate - sampling rate of the request
+// Output : Returns true to keep mixing, false to delete this mixer
+//-----------------------------------------------------------------------------
+bool CAudioMixerWave::SkipSamples( IAudioDevice *pDevice, channel_t *pChannel, 
+	int startSample, int sampleCount, int outputRate, bool forward /*= true*/ )
+{
+	int offset = 0;
+
+	int inputSampleRate = (int)(pChannel->pitch * m_pData->Source().SampleRate());
+	float rate = (float)inputSampleRate / outputRate;
+
+	sampleCount = min( sampleCount, pDevice->PaintBufferSampleCount() );
+
+	int startpos = startSample;
+
+	if ( !forward )
+	{
+		int requestedstart = startSample - (int)( sampleCount * rate );
+		if ( requestedstart < 0 )
+			return false;
+
+		startpos = max( 0, requestedstart );
+		SetSamplePosition( startpos );
+	}
+
+	while ( sampleCount > 0 )
+	{
+		int availableSamples;
+		int inputSampleCount;
+		char *pData = NULL;
+		int outputSampleCount = sampleCount;
+		
+		if ( outputRate != inputSampleRate )
+		{
+			inputSampleCount = (int)(sampleCount * rate);
+			if ( !forward )
+			{
+				startSample = max( 0, startSample - inputSampleCount );
+			}
+			int availableSamples = GetOutputData( (void **)&pData, startSample, inputSampleCount, forward );
+			if ( !availableSamples )
+				break;
+
+			if ( availableSamples < inputSampleCount )
+			{
+				outputSampleCount = (int)(availableSamples / rate);
+				inputSampleCount = availableSamples;
+			}
+
+			// compute new fraction part of sample index
+			float offset = (m_fracOffset / FIX_SCALE) + (rate * outputSampleCount);
+			offset = offset - (float)((int)offset);
+			m_fracOffset = FIX_FLOAT(offset);
+		}
+		else
+		{
+			if ( !forward )
+			{
+				startSample = max( 0, startSample - sampleCount );
+			}
+			availableSamples = GetOutputData( (void **)&pData, startSample, sampleCount, forward );
+			if ( !availableSamples )
+				break;
+			outputSampleCount = availableSamples;
+			inputSampleCount = availableSamples;
+
+		}
+		offset += outputSampleCount;
+		sampleCount -= outputSampleCount;
+		if ( forward )
+		{
+			m_sample += inputSampleCount;
+			m_absoluteSample += inputSampleCount;
+		}
+
+		if ( m_loop != 0 && m_sample >= m_loop )
+		{
+			SetSamplePosition( m_startpos );
+		}
+
+	}
+
+	if ( sampleCount > 0 )
+		return false;
+
+	return true;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: The device calls this to request data.  The mixer must provide the
+//			full amount of samples or have silence in its output stream.
+// Input  : *pDevice - requesting device
+//			sampleCount - number of samples at the output rate
+//			outputRate - sampling rate of the request
+// Output : Returns true to keep mixing, false to delete this mixer
+//-----------------------------------------------------------------------------
+bool CAudioMixerWave::MixDataToDevice( IAudioDevice *pDevice, channel_t *pChannel, int startSample, int sampleCount, int outputRate, bool forward /*= true*/ )
+{
+	int offset = 0;
+
+	int inputSampleRate = (int)(pChannel->pitch * m_pData->Source().SampleRate());
+	float rate = (float)inputSampleRate / outputRate;
+	fixedint fracstep = FIX_FLOAT( rate );
+
+	sampleCount = min( sampleCount, pDevice->PaintBufferSampleCount() );
+
+	int startpos = startSample;
+
+	if ( !forward )
+	{
+		int requestedstart = startSample - (int)( sampleCount * rate );
+		if ( requestedstart < 0 )
+			return false;
+
+		startpos = max( 0, requestedstart );
+		SetSamplePosition( startpos );
+	}
+
+	while ( sampleCount > 0 )
+	{
+		int availableSamples;
+		int inputSampleCount;
+		char *pData = NULL;
+		int outputSampleCount = sampleCount;
+		
+		
+		if ( outputRate != inputSampleRate )
+		{
+			inputSampleCount = (int)(sampleCount * rate);
+
+			int availableSamples = GetOutputData( (void **)&pData, startpos, inputSampleCount, forward );
+			if ( !availableSamples )
+				break;
+
+			if ( availableSamples < inputSampleCount )
+			{
+				outputSampleCount = (int)(availableSamples / rate);
+				inputSampleCount = availableSamples;
+			}
+
+			Mix( pDevice, pChannel, pData, offset, m_fracOffset, fracstep, outputSampleCount, 0, forward );
+			
+			// compute new fraction part of sample index
+			float offset = (m_fracOffset / FIX_SCALE) + (rate * outputSampleCount);
+			offset = offset - (float)((int)offset);
+			m_fracOffset = FIX_FLOAT(offset);
+		}
+		else
+		{
+			availableSamples = GetOutputData( (void **)&pData, startpos, sampleCount, forward );
+			if ( !availableSamples )
+				break;
+
+			outputSampleCount = availableSamples;
+			inputSampleCount = availableSamples;
+
+			Mix( pDevice, pChannel, pData, offset, m_fracOffset, FIX(1), outputSampleCount, 0, forward );
+		}
+		offset += outputSampleCount;
+		sampleCount -= outputSampleCount;
+
+		if ( forward )
+		{
+			m_sample += inputSampleCount;
+			m_absoluteSample += inputSampleCount;
+		}
+
+		if ( m_loop != 0 && m_sample >= m_loop )
+		{
+			SetSamplePosition( m_startpos );
+		}
+
+	}
+
+	if ( sampleCount > 0 )
+		return false;
+
+	return true;
+}

soundsystem/snd_wave_mixer.h

@@ -0,0 +1,31 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $Workfile:     $
+// $Date:         $
+// $NoKeywords: $
+//===========================================================================//
+
+#ifndef SND_WAVE_MIXER_H
+#define SND_WAVE_MIXER_H
+
+#ifdef _WIN32
+#pragma once
+#endif
+
+
+//-----------------------------------------------------------------------------
+// Forward declarations
+//-----------------------------------------------------------------------------
+class CWaveData;
+class CAudioMixer;
+
+
+//-----------------------------------------------------------------------------
+// Wave mixer
+//-----------------------------------------------------------------------------
+CAudioMixer *CreateWaveMixer( CWaveData *data, int format, int channels, int bits );
+
+
+#endif // SND_WAVE_MIXER_H

soundsystem/snd_wave_mixer_adpcm.cpp

@@ -0,0 +1,516 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//=============================================================================//
+#define WIN32_LEAN_AND_MEAN
+#include <windows.h>
+
+#pragma warning( disable: 4201 )
+#include <mmsystem.h>
+#pragma warning( default: 4201 )
+
+#include <mmreg.h>
+#include "snd_wave_source.h"
+#include "snd_wave_mixer_adpcm.h"
+#include "snd_wave_mixer_private.h"
+#include "soundsystem.h"
+
+// max size of ADPCM block in bytes
+#define MAX_BLOCK_SIZE	4096
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Mixer for ADPCM encoded audio
+//-----------------------------------------------------------------------------
+class CAudioMixerWaveADPCM : public CAudioMixerWave
+{
+public:
+	CAudioMixerWaveADPCM( CWaveData *data );
+	~CAudioMixerWaveADPCM( void );
+	
+	virtual void Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress, bool forward = true );
+	virtual int	 GetOutputData( void **pData, int samplePosition, int sampleCount, bool forward = true );
+
+	virtual bool SetSamplePosition( int position, bool scrubbing = false );
+
+private:
+	bool					DecodeBlock( void );
+	int						NumChannels( void );
+	void					DecompressBlockMono( short *pOut, const char *pIn, int count );
+	void					DecompressBlockStereo( short *pOut, const char *pIn, int count );
+
+	void					SetCurrentBlock( int block );
+	int						GetCurrentBlock( void ) const;
+	int						GetBlockNumberForSample( int samplePosition );
+	bool					IsSampleInCurrentBlock( int samplePosition );
+	int						GetFirstSampleForBlock( int blocknum ) const;
+
+	const ADPCMWAVEFORMAT	*m_pFormat;
+	const ADPCMCOEFSET		*m_pCoefficients;
+
+	short					*m_pSamples;
+	int						m_sampleCount;
+	int						m_samplePosition;
+
+	int						m_blockSize;
+	int						m_offset;
+
+	int						m_currentBlock;
+};
+
+
+CAudioMixerWaveADPCM::CAudioMixerWaveADPCM( CWaveData *data ) : CAudioMixerWave( data ) 
+{
+	m_currentBlock = -1;
+	m_pSamples = NULL;
+	m_sampleCount = 0;
+	m_samplePosition = 0;
+	m_offset = 0;
+
+	m_pFormat = (const ADPCMWAVEFORMAT *)m_pData->Source().GetHeader();
+	if ( m_pFormat )
+	{
+		m_pCoefficients = (ADPCMCOEFSET *)((char *)m_pFormat + sizeof(WAVEFORMATEX) + 4);
+
+		// create the decode buffer
+		m_pSamples = new short[m_pFormat->wSamplesPerBlock * m_pFormat->wfx.nChannels];
+
+		// number of bytes for samples
+		m_blockSize = ((m_pFormat->wSamplesPerBlock - 2) * m_pFormat->wfx.nChannels ) / 2;
+		// size of channel header
+		m_blockSize += 7 * m_pFormat->wfx.nChannels;
+//		Assert(m_blockSize < MAX_BLOCK_SIZE);
+	}
+}
+
+
+CAudioMixerWaveADPCM::~CAudioMixerWaveADPCM( void )
+{
+	delete[] m_pSamples;
+}
+
+
+int	CAudioMixerWaveADPCM::NumChannels( void )
+{
+	if ( m_pFormat )
+	{
+		return m_pFormat->wfx.nChannels;
+	}
+	return 0;
+}
+
+void CAudioMixerWaveADPCM::Mix( IAudioDevice *pDevice, channel_t *pChannel, void *pData, int outputOffset, int inputOffset, fixedint fracRate, int outCount, int timecompress, bool forward /*= true*/ )
+{
+	if ( NumChannels() == 1 )
+		pDevice->Mix16Mono( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
+	else
+		pDevice->Mix16Stereo( pChannel, (short *)pData, outputOffset, inputOffset, fracRate, outCount, timecompress, forward );
+}
+
+
+static int error_sign_lut[] =		  { 0, 1, 2, 3, 4, 5, 6, 7, -8, -7, -6, -5, -4, -3, -2, -1 };
+static int error_coefficients_lut[] = { 230, 230, 230, 230, 307, 409, 512, 614,
+										768, 614, 512, 409, 307, 230, 230, 230 };
+
+//-----------------------------------------------------------------------------
+// Purpose: ADPCM decompress a single block of 1-channel audio
+// Input  : *pOut - output buffer 16-bit
+//			*pIn - input block
+//			count - number of samples to decode (to support partial blocks)
+//-----------------------------------------------------------------------------
+void CAudioMixerWaveADPCM::DecompressBlockMono( short *pOut, const char *pIn, int count )
+{
+
+	int pred = *pIn++;
+	int co1 = m_pCoefficients[pred].iCoef1;
+	int co2 = m_pCoefficients[pred].iCoef2;
+
+	// read initial delta
+	int delta = *((short *)pIn);
+	pIn += 2;
+
+	// read initial samples for prediction
+	int samp1 = *((short *)pIn);
+	pIn += 2;
+
+	int samp2 = *((short *)pIn);
+	pIn += 2;
+
+	// write out the initial samples (stored in reverse order)
+	*pOut++ = (short)samp2;
+	*pOut++ = (short)samp1;
+
+	// subtract the 2 samples in the header
+	count -= 2;
+
+	// this is a toggle to read nibbles, first nibble is high
+	int high = 1;
+
+	int error = 0, sample = 0;
+
+	// now process the block
+	while ( count )
+	{
+		// read the error nibble from the input stream
+		if ( high )
+		{
+			sample = (unsigned char) (*pIn++);
+			// high nibble
+			error = sample >> 4;
+			// cache low nibble for next read
+			sample = sample & 0xf;
+			// Next read is from cache, not stream
+			high = 0;
+		}
+		else
+		{
+			// stored in previous read (low nibble)
+			error = sample;
+			// next read is from stream
+			high = 1;
+		}
+		// convert to signed with LUT
+		int errorSign = error_sign_lut[error];
+
+		// interpolate the new sample
+		int predSample = (samp1 * co1) + (samp2 * co2);
+		// coefficients are fixed point 8-bit, so shift back to 16-bit integer
+		predSample >>= 8;
+
+		// Add in current error estimate
+		predSample += (errorSign * delta);
+
+		// Correct error estimate
+		delta = (delta * error_coefficients_lut[error]) >> 8;
+		// Clamp error estimate
+		if ( delta < 16 )
+			delta = 16;
+
+		// clamp
+		if ( predSample > 32767L )
+			predSample = 32767L;
+		else if ( predSample < -32768L )
+			predSample = -32768L;
+		
+		// output
+		*pOut++ = (short)predSample;
+		// move samples over
+		samp2 = samp1;
+		samp1 = predSample;
+
+		count--;
+	}
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Decode a single block of stereo ADPCM audio
+// Input  : *pOut - 16-bit output buffer
+//			*pIn - ADPCM encoded block data
+//			count - number of sample pairs to decode
+//-----------------------------------------------------------------------------
+void CAudioMixerWaveADPCM::DecompressBlockStereo( short *pOut, const char *pIn, int count )
+{
+	int pred[2], co1[2], co2[2];
+	int i;
+
+	for ( i = 0; i < 2; i++ )
+	{
+		pred[i] = *pIn++;
+		co1[i] = m_pCoefficients[pred[i]].iCoef1;
+		co2[i] = m_pCoefficients[pred[i]].iCoef2;
+	}
+
+	int delta[2], samp1[2], samp2[2];
+
+	for ( i = 0; i < 2; i++, pIn += 2 )
+	{
+		// read initial delta
+		delta[i] = *((short *)pIn);
+	}
+
+	// read initial samples for prediction
+	for ( i = 0; i < 2; i++, pIn += 2 )
+	{
+		samp1[i] = *((short *)pIn);
+	}
+	for ( i = 0; i < 2; i++, pIn += 2 )
+	{
+		samp2[i] = *((short *)pIn);
+	}
+
+	// write out the initial samples (stored in reverse order)
+	*pOut++ = (short)samp2[0];	// left
+	*pOut++ = (short)samp2[1];	// right
+	*pOut++ = (short)samp1[0];	// left
+	*pOut++ = (short)samp1[1];	// right
+
+	// subtract the 2 samples in the header
+	count -= 2;
+
+	// this is a toggle to read nibbles, first nibble is high
+	int high = 1;
+
+	int error, sample = 0;
+
+	// now process the block
+	while ( count )
+	{
+		for ( i = 0; i < 2; i++ )
+		{
+			// read the error nibble from the input stream
+			if ( high )
+			{
+				sample = (unsigned char) (*pIn++);
+				// high nibble
+				error = sample >> 4;
+				// cache low nibble for next read
+				sample = sample & 0xf;
+				// Next read is from cache, not stream
+				high = 0;
+			}
+			else
+			{
+				// stored in previous read (low nibble)
+				error = sample;
+				// next read is from stream
+				high = 1;
+			}
+			// convert to signed with LUT
+			int errorSign = error_sign_lut[error];
+
+			// interpolate the new sample
+			int predSample = (samp1[i] * co1[i]) + (samp2[i] * co2[i]);
+			// coefficients are fixed point 8-bit, so shift back to 16-bit integer
+			predSample >>= 8;
+
+			// Add in current error estimate
+			predSample += (errorSign * delta[i]);
+
+			// Correct error estimate
+			delta[i] = (delta[i] * error_coefficients_lut[error]) >> 8;
+			// Clamp error estimate
+			if ( delta[i] < 16 )
+				delta[i] = 16;
+
+			// clamp
+			if ( predSample > 32767L )
+				predSample = 32767L;
+			else if ( predSample < -32768L )
+				predSample = -32768L;
+			
+			// output
+			*pOut++ = (short)predSample;
+			// move samples over
+			samp2[i] = samp1[i];
+			samp1[i] = predSample;
+		}
+		count--;
+	}
+}
+
+
+bool CAudioMixerWaveADPCM::DecodeBlock( void )
+{
+	char tmpBlock[MAX_BLOCK_SIZE];
+	char *pData;
+
+	int available = m_pData->ReadSourceData( (void **) (&pData), m_offset, m_blockSize );
+	if ( available < m_blockSize )
+	{
+		int total = 0;
+		while ( available && total < m_blockSize )
+		{
+			memcpy( tmpBlock + total, pData, available );
+			total += available;
+			available = m_pData->ReadSourceData( (void **) (&pData), m_offset + total, m_blockSize - total );
+		}
+		pData = tmpBlock;
+		available = total;
+	}
+
+	Assert( m_blockSize > 0 );
+
+	// Current block number is based on starting offset
+	int blockNumber = m_offset / m_blockSize;
+	SetCurrentBlock( blockNumber );
+
+	if ( !available )
+	{
+		return false;
+	}
+
+	// advance the file pointer
+	m_offset += available;
+
+	int channelCount = NumChannels();
+
+	// this is sample pairs for stereo, samples for mono
+	m_sampleCount = m_pFormat->wSamplesPerBlock;
+
+	// short block?, fixup sample count (2 samples per byte, divided by number of channels per sample set)
+	m_sampleCount -= ((m_blockSize - available) * 2) / channelCount;
+
+	// new block, start at the first sample
+	m_samplePosition = 0;
+
+	// no need to subclass for different channel counts...
+	if ( channelCount == 1 )
+	{
+		DecompressBlockMono( m_pSamples, pData, m_sampleCount );
+	}
+	else
+	{
+		DecompressBlockStereo( m_pSamples, pData, m_sampleCount );
+	}
+	return true;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : block - 
+//-----------------------------------------------------------------------------
+void CAudioMixerWaveADPCM::SetCurrentBlock( int block )
+{
+	m_currentBlock = block;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Output : int
+//-----------------------------------------------------------------------------
+int CAudioMixerWaveADPCM::GetCurrentBlock( void ) const
+{
+	return m_currentBlock;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : samplePosition - 
+// Output : int
+//-----------------------------------------------------------------------------
+int CAudioMixerWaveADPCM::GetBlockNumberForSample( int samplePosition )
+{
+	int blockNum = samplePosition / m_pFormat->wSamplesPerBlock;
+	return blockNum;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : samplePosition - 
+// Output : Returns true on success, false on failure.
+//-----------------------------------------------------------------------------
+bool CAudioMixerWaveADPCM::IsSampleInCurrentBlock( int samplePosition )
+{
+	int currentBlock = GetCurrentBlock();
+
+	int startSample = currentBlock * m_pFormat->wSamplesPerBlock;
+	int endSample = startSample + m_pFormat->wSamplesPerBlock - 1;
+
+	if ( samplePosition >= startSample &&
+		 samplePosition <= endSample )
+	{
+		return true;
+	}
+
+	return false;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : blocknum - 
+// Output : int
+//-----------------------------------------------------------------------------
+int CAudioMixerWaveADPCM::GetFirstSampleForBlock( int blocknum ) const
+{
+	return m_pFormat->wSamplesPerBlock * blocknum;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Read existing buffer or decompress a new block when necessary
+// Input  : **pData - output data pointer
+//			sampleCount - number of samples (or pairs)
+// Output : int - available samples (zero to stop decoding)
+//-----------------------------------------------------------------------------
+int CAudioMixerWaveADPCM::GetOutputData( void **pData, int samplePosition, int sampleCount, bool forward /*= true*/ )
+{
+	int requestedBlock = GetBlockNumberForSample( samplePosition );
+	if ( requestedBlock != GetCurrentBlock() )
+	{
+		// Ran out of data!!!
+		if ( !SetSamplePosition( samplePosition ) )
+			return 0;
+	}
+
+	Assert( requestedBlock == GetCurrentBlock() );
+
+	if ( m_samplePosition >= m_sampleCount )
+	{
+		if ( !DecodeBlock() )
+			return 0;
+	}
+
+	if ( m_samplePosition < m_sampleCount )
+	{
+		*pData = (void *)(m_pSamples + m_samplePosition * NumChannels());
+		int available = m_sampleCount - m_samplePosition;
+		if ( available > sampleCount )
+			available = sampleCount;
+
+		m_samplePosition += available;
+		return available;
+	}
+
+	return 0;
+}
+
+
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : position - 
+//-----------------------------------------------------------------------------
+bool CAudioMixerWaveADPCM::SetSamplePosition( int position, bool scrubbing )
+{
+	position = max( 0, position );
+
+	CAudioMixerWave::SetSamplePosition( position, scrubbing );
+
+	int requestedBlock	= GetBlockNumberForSample( position );
+	int firstSample		= GetFirstSampleForBlock( requestedBlock );
+
+	if ( firstSample >= m_pData->Source().SampleCount() )
+	{
+		// Read past end of file!!!
+		return false;
+	}
+
+	int currentSample = ( position - firstSample );
+
+	if ( requestedBlock != GetCurrentBlock() )
+	{
+		// Rewind file to beginning of block
+		m_offset = requestedBlock * m_blockSize;
+		if ( !DecodeBlock() )
+		{
+			return false;
+		}
+	}
+	
+	m_samplePosition = currentSample;
+	return true;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Abstract factory function for ADPCM mixers
+// Input  : *data - wave data access object
+//			channels - 
+// Output : CAudioMixer
+//-----------------------------------------------------------------------------
+CAudioMixer *CreateADPCMMixer( CWaveData *data )
+{
+	return new CAudioMixerWaveADPCM( data );
+}

soundsystem/snd_wave_mixer_adpcm.h

@@ -0,0 +1,24 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $Workfile:     $
+// $Date:         $
+//
+//-----------------------------------------------------------------------------
+// $Log: $
+//
+// $NoKeywords: $
+//=============================================================================//
+
+#ifndef SND_WAVE_MIXER_ADPCM_H
+#define SND_WAVE_MIXER_ADPCM_H
+#pragma once
+
+
+class CAudioMixer;
+class CWaveData;
+
+CAudioMixer *CreateADPCMMixer( CWaveData *data );
+
+#endif // SND_WAVE_MIXER_ADPCM_H

soundsystem/snd_wave_mixer_private.h

@@ -0,0 +1,103 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $Workfile:     $
+// $Date:         $
+// $NoKeywords: $
+//===========================================================================//
+
+#ifndef SND_WAVE_MIXER_PRIVATE_H
+#define SND_WAVE_MIXER_PRIVATE_H
+
+#ifdef _WIN32
+#pragma once
+#endif
+
+#include "soundsystem/snd_audio_source.h"
+#include "soundsystem/snd_device.h"
+#include "snd_wave_mixer.h"
+
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Linear iterator over source data.
+//			Keeps track of position in source, and maintains necessary buffers
+//-----------------------------------------------------------------------------
+class CWaveData
+{
+public:
+	virtual ~CWaveData( void ) {}
+	virtual CAudioSourceWave &Source( void ) = 0;
+	virtual int ReadSourceData( void **pData, int sampleIndex, int sampleCount, bool forward = true ) = 0;
+};
+
+class CAudioMixerWave : public CAudioMixer
+{
+public:
+	CAudioMixerWave( CWaveData *data );
+	virtual ~CAudioMixerWave( void );
+
+	virtual bool MixDataToDevice( IAudioDevice *pDevice, channel_t *pChannel, int startSample, int sampleCount, int outputRate, bool forward = true );
+	virtual void IncrementSamples( channel_t *pChannel, int startSample, int sampleCount,int outputRate, bool forward = true );
+	virtual bool SkipSamples( IAudioDevice *pDevice, channel_t *pChannel, int startSample, int sampleCount,int outputRate, bool forward = true  );
+	virtual void Mix( IAudioDevice *pDevice, 
+					channel_t *pChannel, 
+					void *pData, 
+					int outputOffset, 
+					int inputOffset, 
+					fixedint fracRate, 
+					int outCount, 
+					int timecompress,
+					bool forward = true ) = 0;
+
+	virtual int	 GetOutputData( void **pData, int samplePosition, int sampleCount, bool forward = true );
+
+	virtual CAudioSource *GetSource( void );
+
+	virtual int GetSamplePosition( void );
+	virtual int GetScubPosition( void );
+
+	virtual bool SetSamplePosition( int position, bool scrubbing = false );
+	virtual void SetLoopPosition( int position );
+	virtual int	GetStartPosition( void );
+
+	virtual bool	GetActive( void );
+	virtual void	SetActive( bool active );
+
+	virtual void	SetModelIndex( int index );
+	virtual int		GetModelIndex( void ) const;
+
+	virtual void	SetDirection( bool forward );
+	virtual bool	GetDirection( void ) const;
+
+	virtual void	SetAutoDelete( bool autodelete );
+	virtual bool	GetAutoDelete( void ) const;
+
+	virtual void	SetVolume( float volume );
+	virtual channel_t *GetChannel();
+
+protected:
+	int					m_sample;
+	int					m_absoluteSample;
+	int					m_scubSample;
+	int					m_startpos;
+	int					m_loop;
+	int					m_fracOffset;
+	CWaveData			*m_pData;
+
+	int					m_absoluteStartPos;
+
+	bool				m_bActive;
+	// Associated playback model in faceposer
+	int					m_nModelIndex;
+
+	bool				m_bForward;
+
+	bool				m_bAutoDelete;
+
+	channel_t			*m_pChannel;
+};
+
+
+#endif // SND_WAVE_MIXER_PRIVATE_H

soundsystem/snd_wave_source.cpp

@@ -0,0 +1,602 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $NoKeywords: $
+//
+//===========================================================================//
+#include <stdio.h>
+#include <string.h>
+#include <stdlib.h>
+#include <windows.h>
+#include "tier2/riff.h"
+#include "snd_wave_source.h"
+#include "snd_wave_mixer_private.h"
+#include "soundsystem/snd_audio_source.h"
+#include <mmsystem.h>		// wave format
+#include <mmreg.h>			// adpcm format
+#include "soundsystem.h"
+#include "filesystem.h"
+#include "tier1/utlbuffer.h"
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Implements the RIFF i/o interface on stdio
+//-----------------------------------------------------------------------------
+class StdIOReadBinary : public IFileReadBinary
+{
+public:
+	int open( const char *pFileName )
+	{
+		return (int)g_pFullFileSystem->Open( pFileName, "rb", "GAME" );
+	}
+
+	int read( void *pOutput, int size, int file )
+	{
+		if ( !file )
+			return 0;
+
+		return g_pFullFileSystem->Read( pOutput, size, (FileHandle_t)file );
+	}
+
+	void seek( int file, int pos )
+	{
+		if ( !file )
+			return;
+
+		g_pFullFileSystem->Seek( (FileHandle_t)file, pos, FILESYSTEM_SEEK_HEAD );
+	}
+
+	unsigned int tell( int file )
+	{
+		if ( !file )
+			return 0;
+
+		return g_pFullFileSystem->Tell( (FileHandle_t)file );
+	}
+
+	unsigned int size( int file )
+	{
+		if ( !file )
+			return 0;
+
+		return g_pFullFileSystem->Size( (FileHandle_t)file );
+	}
+
+	void close( int file )
+	{
+		if ( !file )
+			return;
+
+		g_pFullFileSystem->Close( (FileHandle_t)file );
+	}
+};
+
+static StdIOReadBinary io;
+
+#define RIFF_WAVE			MAKEID('W','A','V','E')
+#define WAVE_FMT			MAKEID('f','m','t',' ')
+#define WAVE_DATA			MAKEID('d','a','t','a')
+#define WAVE_FACT			MAKEID('f','a','c','t')
+#define WAVE_CUE			MAKEID('c','u','e',' ')
+
+void ChunkError( unsigned int id )
+{
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Init to empty wave
+//-----------------------------------------------------------------------------
+CAudioSourceWave::CAudioSourceWave( void )
+{
+	m_bits = 0;
+	m_rate = 0;
+	m_channels = 0;
+	m_format = 0;
+	m_pHeader = NULL;
+	// no looping
+	m_loopStart = -1;
+	m_sampleSize = 1;
+	m_sampleCount = 0;
+}
+
+
+CAudioSourceWave::~CAudioSourceWave( void )
+{
+	// for non-standard waves, we store a copy of the header in RAM
+	delete[] m_pHeader;
+	//	m_pWords points into m_pWordBuffer, no need to delete
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Init the wave data.
+// Input  : *pHeaderBuffer - the RIFF fmt chunk
+//			headerSize - size of that chunk
+//-----------------------------------------------------------------------------
+void CAudioSourceWave::Init( const char *pHeaderBuffer, int headerSize )
+{
+	const WAVEFORMATEX *pHeader = (const WAVEFORMATEX *)pHeaderBuffer;
+
+	// copy the relevant header data
+	m_format = pHeader->wFormatTag;
+	m_bits = pHeader->wBitsPerSample;
+	m_rate = pHeader->nSamplesPerSec;
+	m_channels = pHeader->nChannels;
+
+	m_sampleSize = (m_bits * m_channels) / 8;
+	
+	// this can never be zero -- other functions divide by this. 
+	// This should never happen, but avoid crashing
+	if ( m_sampleSize <= 0 )
+		m_sampleSize = 1;
+
+	// For non-standard waves (like ADPCM) store the header, it has some useful data
+	if ( m_format != WAVE_FORMAT_PCM )
+	{
+		m_pHeader = new char[headerSize];
+		memcpy( m_pHeader, pHeader, headerSize );
+		if ( m_format == WAVE_FORMAT_ADPCM )
+		{
+			// treat ADPCM sources as a file of bytes.  They are decoded by the mixer
+			m_sampleSize = 1;
+		}
+	}
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Output : float
+//-----------------------------------------------------------------------------
+float CAudioSourceWave::TrueSampleSize( void )
+{
+	if ( m_format == WAVE_FORMAT_ADPCM )
+	{
+		return 0.5f;
+	}
+	return (float)m_sampleSize;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Total number of samples in this source
+// Output : int
+//-----------------------------------------------------------------------------
+int CAudioSourceWave::SampleCount( void ) 
+{
+	if ( m_format == WAVE_FORMAT_ADPCM )
+	{
+		ADPCMWAVEFORMAT *pFormat = (ADPCMWAVEFORMAT *)m_pHeader;
+		int blockSize = ((pFormat->wSamplesPerBlock - 2) * pFormat->wfx.nChannels ) / 2;
+		blockSize += 7 * pFormat->wfx.nChannels;
+
+		int blockCount = m_sampleCount / blockSize;
+		int blockRem = m_sampleCount % blockSize;
+		
+		// total samples in complete blocks
+		int sampleCount = blockCount * pFormat->wSamplesPerBlock;
+
+		// add remaining in a short block
+		if ( blockRem )
+		{
+			sampleCount += pFormat->wSamplesPerBlock - (((blockSize - blockRem) * 2) / m_channels);
+		}
+		return sampleCount;
+	}
+	return m_sampleCount; 
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Do any sample conversion
+//			For 8 bit PCM, convert to signed because the mixing routine assumes this
+// Input  : *pData - pointer to sample data
+//			sampleCount - number of samples
+//-----------------------------------------------------------------------------
+void CAudioSourceWave::ConvertSamples( char *pData, int sampleCount )
+{
+	if ( m_format == WAVE_FORMAT_PCM )
+	{
+		if ( m_bits == 8 )
+		{
+			for ( int i = 0; i < sampleCount; i++ )
+			{
+				for ( int j = 0; j < m_channels; j++ )
+				{
+					*pData = (unsigned char)((int)((unsigned)*pData) - 128);
+					pData++;
+				}
+			}
+		}
+	}
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : &walk - 
+//-----------------------------------------------------------------------------
+void CAudioSourceWave::ParseSentence( IterateRIFF &walk )
+{
+	CUtlBuffer buf( 0, 0, CUtlBuffer::TEXT_BUFFER );
+
+	buf.EnsureCapacity( walk.ChunkSize() );
+	walk.ChunkRead( buf.Base() );
+	buf.SeekPut( CUtlBuffer::SEEK_HEAD, walk.ChunkSize() );
+
+	m_Sentence.InitFromDataChunk( buf.Base(), buf.TellPut() );
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Parse base chunks
+// Input  : &walk - riff file to parse
+//		  : chunkName - name of the chunk to parse
+//-----------------------------------------------------------------------------
+// UNDONE: Move parsing loop here and drop each chunk into a virtual function
+//			instead of this being virtual.
+void CAudioSourceWave::ParseChunk( IterateRIFF &walk, int chunkName )
+{
+	switch( chunkName )
+	{
+		case WAVE_CUE:
+			{
+				m_loopStart = ParseCueChunk( walk );
+			}
+			break;
+		case WAVE_VALVEDATA:
+			{
+				ParseSentence( walk );
+			}
+			break;
+			// unknown/don't care
+		default:
+			{
+				ChunkError( walk.ChunkName() );
+			}
+			break;
+	}
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Output : CSentence
+//-----------------------------------------------------------------------------
+CSentence *CAudioSourceWave::GetSentence( void )
+{
+	return &m_Sentence;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Bastardized construction routine.  This is just to avoid complex
+//			constructor functions so code can be shared more easily by sub-classes
+// Input  : *pFormatBuffer - RIFF header
+//			formatSize - header size
+//			&walk - RIFF file
+//-----------------------------------------------------------------------------
+void CAudioSourceWave::Setup( const char *pFormatBuffer, int formatSize, IterateRIFF &walk )
+{
+	Init( pFormatBuffer, formatSize );
+
+	while ( walk.ChunkAvailable() )
+	{
+		ParseChunk( walk, walk.ChunkName() );
+		walk.ChunkNext();
+	}
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Wave file that is completely in memory
+//			UNDONE: Implement Lock/Unlock and caching
+//-----------------------------------------------------------------------------
+class CAudioSourceMemWave : public CAudioSourceWave
+{
+public:
+	CAudioSourceMemWave( void );
+	~CAudioSourceMemWave( void );
+
+	// Create an instance (mixer) of this audio source
+	virtual CAudioMixer		*CreateMixer( void );
+
+	virtual void			ParseChunk( IterateRIFF &walk, int chunkName );
+	void					ParseDataChunk( IterateRIFF &walk );
+
+	virtual int				GetOutputData( void **pData, int samplePosition, int sampleCount, bool forward = true );
+	virtual float			GetRunningLength( void ) { return CAudioSourceWave::GetRunningLength(); };
+
+	virtual int				GetNumChannels();
+
+private:
+	char	*m_pData;		// wave data
+};
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Iterator for wave data (this is to abstract streaming/buffering)
+//-----------------------------------------------------------------------------
+class CWaveDataMemory : public CWaveData
+{
+public:
+	CWaveDataMemory( CAudioSourceWave &source ) : m_source(source) {}
+	~CWaveDataMemory( void ) {}
+	CAudioSourceWave &Source( void ) { return m_source; }
+	
+	// this file is in memory, simply pass along the data request to the source
+	virtual int ReadSourceData( void **pData, int sampleIndex, int sampleCount, bool forward /*= true*/ )
+	{
+		return m_source.GetOutputData( pData, sampleIndex, sampleCount, forward );
+	}
+private:
+	CAudioSourceWave		&m_source;	// pointer to source
+};
+
+
+//-----------------------------------------------------------------------------
+// Purpose: NULL the wave data pointer (we haven't loaded yet)
+//-----------------------------------------------------------------------------
+CAudioSourceMemWave::CAudioSourceMemWave( void )
+{
+	m_pData = NULL;
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: Free any wave data we've allocated
+//-----------------------------------------------------------------------------
+CAudioSourceMemWave::~CAudioSourceMemWave( void )
+{
+	delete[] m_pData;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Creates a mixer and initializes it with an appropriate mixer
+//-----------------------------------------------------------------------------
+CAudioMixer *CAudioSourceMemWave::CreateMixer( void )
+{
+	return CreateWaveMixer( new CWaveDataMemory(*this), m_format, m_channels, m_bits );
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: parse chunks with unique processing to in-memory waves
+// Input  : &walk - RIFF file
+//-----------------------------------------------------------------------------
+void CAudioSourceMemWave::ParseChunk( IterateRIFF &walk, int chunkName )
+{
+	switch( chunkName )
+	{
+		// this is the audio data
+	case WAVE_DATA:
+		{
+			ParseDataChunk( walk );
+		}
+		return;
+	}
+
+	CAudioSourceWave::ParseChunk( walk, chunkName );
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: reads the actual sample data and parses it
+// Input  : &walk - RIFF file
+//-----------------------------------------------------------------------------
+void CAudioSourceMemWave::ParseDataChunk( IterateRIFF &walk )
+{
+	int size = walk.ChunkSize();
+	
+	// create a buffer for the samples
+	m_pData = new char[size];
+
+	// load them into memory
+	walk.ChunkRead( m_pData );
+
+	if ( m_format == WAVE_FORMAT_PCM )
+	{
+		// number of samples loaded
+		m_sampleCount = size / m_sampleSize;
+
+		// some samples need to be converted
+		ConvertSamples( m_pData, m_sampleCount );
+	}
+	else if ( m_format == WAVE_FORMAT_ADPCM )
+	{
+		// The ADPCM mixers treat the wave source as a flat file of bytes.
+		m_sampleSize = 1;
+		// Since each "sample" is a byte (this is a flat file), the number of samples is the file size
+		m_sampleCount = size;
+
+		// file says 4, output is 16
+		m_bits = 16;
+	}
+}
+
+int CAudioSourceMemWave::GetNumChannels()
+{
+	return m_channels;
+}
+
+
+
+//-----------------------------------------------------------------------------
+// Purpose: parses loop information from a cue chunk
+// Input  : &walk - RIFF iterator
+// Output : int loop start position
+//-----------------------------------------------------------------------------
+int CAudioSourceWave::ParseCueChunk( IterateRIFF &walk )
+{
+	// Cue chunk as specified by RIFF format
+	// see $/research/jay/sound/riffnew.htm
+	struct 
+	{
+		unsigned int dwName; 
+		unsigned int dwPosition;
+		unsigned int fccChunk;
+		unsigned int dwChunkStart;
+		unsigned int dwBlockStart; 
+		unsigned int dwSampleOffset;
+	} cue_chunk;
+
+	int cueCount;
+
+	// assume that the cue chunk stored in the wave is the start of the loop
+	// assume only one cue chunk, UNDONE: Test this assumption here?
+	cueCount = walk.ChunkReadInt();
+
+	walk.ChunkReadPartial( &cue_chunk, sizeof(cue_chunk) );
+	return cue_chunk.dwSampleOffset;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: get the wave header
+//-----------------------------------------------------------------------------
+void *CAudioSourceWave::GetHeader( void )
+{
+	return m_pHeader;
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: wrap the position wrt looping
+// Input  : samplePosition - absolute position
+// Output : int - looped position
+//-----------------------------------------------------------------------------
+int CAudioSourceWave::ConvertLoopedPosition( int samplePosition )
+{
+	// if the wave is looping and we're past the end of the sample
+	// convert to a position within the loop
+	// At the end of the loop, we return a short buffer, and subsequent call
+	// will loop back and get the rest of the buffer
+	if ( m_loopStart >= 0 )
+	{
+		if ( samplePosition >= m_sampleCount )
+		{
+			// size of loop
+			int loopSize = m_sampleCount - m_loopStart;
+			// subtract off starting bit of the wave
+			samplePosition -= m_loopStart;
+			
+			if ( loopSize )
+			{
+				// "real" position in memory (mod off extra loops)
+				samplePosition = m_loopStart + (samplePosition % loopSize);
+			}
+			// ERROR? if no loopSize
+		}
+	}
+
+	return samplePosition;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+// Input  : **pData - output pointer to samples
+//			samplePosition - position (in samples not bytes) 
+//			sampleCount - number of samples (not bytes)
+// Output : int - number of samples available
+//-----------------------------------------------------------------------------
+int CAudioSourceMemWave::GetOutputData( void **pData, int samplePosition, int sampleCount, bool forward /*= true*/ )
+{
+	// handle position looping
+	samplePosition = ConvertLoopedPosition( samplePosition );
+
+	// how many samples are available (linearly not counting looping)
+	int availableSampleCount = m_sampleCount - samplePosition;
+	if ( !forward )
+	{
+		if ( samplePosition >= m_sampleCount )
+		{
+			availableSampleCount = 0;
+		}
+		else
+		{
+			availableSampleCount = samplePosition;
+		}
+	}
+
+	// may be asking for a sample out of range, clip at zero
+	if ( availableSampleCount < 0 )
+		availableSampleCount = 0;
+
+	// clip max output samples to max available
+	if ( sampleCount > availableSampleCount )
+		sampleCount = availableSampleCount;
+
+	// byte offset in sample database
+	samplePosition *= m_sampleSize;
+
+	// if we are returning some samples, store the pointer
+	if ( sampleCount )
+	{
+		*pData = m_pData + samplePosition;
+	}
+
+	return sampleCount;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Create a wave audio source (streaming or in memory)
+// Input  : *pName - file name
+//			streaming - if true, don't load, stream each instance
+// Output : CAudioSource * - a new source
+//-----------------------------------------------------------------------------
+// UNDONE : Pool these and check for duplicates?
+CAudioSource *CreateWave( const char *pName )
+{
+	char formatBuffer[1024];
+	InFileRIFF riff( pName, io );
+
+	if ( riff.RIFFName() != RIFF_WAVE )
+	{
+		Warning("Bad RIFF file type %s\n", pName );
+		return NULL;
+	}
+
+	// set up the iterator for the whole file (root RIFF is a chunk)
+	IterateRIFF walk( riff, riff.RIFFSize() );
+
+	int format = 0;
+	int formatSize = 0;
+
+	// This chunk must be first as it contains the wave's format
+	// break out when we've parsed it
+	while ( walk.ChunkAvailable() && format == 0 )
+	{
+		switch( walk.ChunkName() )
+		{
+		case WAVE_FMT:
+			{
+				if ( walk.ChunkSize() <= 1024 )
+				{
+					walk.ChunkRead( formatBuffer );
+					formatSize = walk.ChunkSize();
+					format = ((WAVEFORMATEX *)formatBuffer)->wFormatTag;
+				}
+			}
+			break;
+		default:
+			{
+				ChunkError( walk.ChunkName() );
+			}
+			break;
+		}
+		walk.ChunkNext();
+	}
+
+	// Not really a WAVE file or no format chunk, bail
+	if ( !format )
+		return NULL;
+
+	CAudioSourceWave *pWave;
+
+	// create the source from this file
+	pWave = new CAudioSourceMemWave();
+
+	// init the wave source
+	pWave->Setup( formatBuffer, formatSize, walk );
+
+	return pWave;
+}
+
+//-----------------------------------------------------------------------------
+// Purpose: Wrapper for CreateWave()
+//-----------------------------------------------------------------------------
+CAudioSource *Audio_CreateMemoryWave( const char *pName )
+{
+	return CreateWave( pName );
+}

soundsystem/snd_wave_source.h

@@ -0,0 +1,76 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: 
+//
+// $Workfile:     $
+// $Date:         $
+// $NoKeywords: $
+//=============================================================================//
+
+#ifndef SND_WAVE_SOURCE_H
+#define SND_WAVE_SOURCE_H
+
+#ifdef _WIN32
+#pragma once
+#endif
+
+#include "soundsystem/snd_audio_source.h"
+#include "sentence.h"
+
+class IterateRIFF;
+
+class CAudioSourceWave : public CAudioSource
+{
+public:
+	CAudioSourceWave( void );
+	~CAudioSourceWave( void );
+
+	void Setup( const char *pFormat, int formatSize, IterateRIFF &walk );
+
+	virtual int				SampleRate( void ) { return m_rate; }
+	inline int				SampleSize( void ) { return m_sampleSize; }
+	virtual float			TrueSampleSize( void );
+
+	void					*GetHeader( void );
+
+	// Legacy
+	virtual	void			ParseChunk( IterateRIFF &walk, int chunkName );
+
+	virtual void			ParseSentence( IterateRIFF &walk );
+	
+	void					ConvertSamples( char *pData, int sampleCount );
+	bool					IsLooped( void ) { return (m_loopStart >= 0) ? true : false; }
+	bool					IsStreaming( void ) { return false; }
+	int						ConvertLoopedPosition( int samplePosition );
+
+	int						SampleCount( void );
+
+	virtual float			GetRunningLength( void )
+							{ 
+								if ( m_rate > 0.0 ) 
+								{
+									return (float)SampleCount() / m_rate;
+								}
+								return 0.0f; }
+		
+	CSentence				*GetSentence( void );
+
+protected:
+	// returns the loop start from a cue chunk
+	int						ParseCueChunk( IterateRIFF &walk );
+	void					Init( const char *pHeaderBuffer, int headerSize );
+
+	int				m_bits;
+	int				m_rate;
+	int				m_channels;
+	int				m_format;
+	int				m_sampleSize;
+	int				m_loopStart;
+	int				m_sampleCount;
+
+private:
+	char			*m_pHeader;
+	CSentence		m_Sentence;
+};
+
+#endif // SND_WAVE_SOURCE_H

soundsystem/soundsystem.cpp

@@ -0,0 +1,279 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: DLL interface for low-level sound utilities
+//
+//===========================================================================//
+
+#include "soundsystem/isoundsystem.h"
+#include "filesystem.h"
+#include "tier1/strtools.h"
+#include "tier1/convar.h"
+#include "mathlib/mathlib.h"
+#include "soundsystem/snd_device.h"
+#include "datacache/idatacache.h"
+#include "soundchars.h"
+#include "tier1/utldict.h"
+#include "snd_wave_source.h"
+#include "snd_dev_wave.h"
+#include "tier2/tier2.h"
+
+
+//-----------------------------------------------------------------------------
+// External interfaces
+//-----------------------------------------------------------------------------
+IAudioDevice *g_pAudioDevice = NULL;
+ISoundSystem *g_pSoundSystem = NULL;
+IDataCache *g_pDataCache = NULL;
+
+
+//-----------------------------------------------------------------------------
+// Globals
+//-----------------------------------------------------------------------------
+int g_nSoundFrameCount = 0;
+
+
+//-----------------------------------------------------------------------------
+// Purpose: DLL interface for low-level sound utilities
+//-----------------------------------------------------------------------------
+class CSoundSystem : public CTier2AppSystem< ISoundSystem >
+{
+	typedef CTier2AppSystem< ISoundSystem > BaseClass;
+
+public:
+	// Inherited from IAppSystem
+	virtual bool Connect( CreateInterfaceFn factory );
+	virtual void Disconnect();
+	virtual void *QueryInterface( const char *pInterfaceName );
+	virtual InitReturnVal_t Init();
+	virtual void Shutdown();
+
+	void		Update( float dt );
+	void		Flush( void );
+
+	CAudioSource *FindOrAddSound( const char *filename );
+	CAudioSource *LoadSound( const char *wavfile );
+	void		PlaySound( CAudioSource *source, float volume, CAudioMixer **ppMixer );
+
+	bool		IsSoundPlaying( CAudioMixer *pMixer );
+	CAudioMixer *FindMixer( CAudioSource *source );
+
+	void		StopAll( void );
+	void		StopSound( CAudioMixer *mixer );
+
+private:
+	struct CSoundFile
+	{
+		char				filename[ 512 ];
+		CAudioSource		*source;
+		long				filetime;
+	};
+
+	IAudioDevice *m_pAudioDevice;
+	float		m_flElapsedTime;
+	CUtlVector < CSoundFile > m_ActiveSounds;
+};
+
+
+//-----------------------------------------------------------------------------
+// Singleton interface
+//-----------------------------------------------------------------------------
+static CSoundSystem s_SoundSystem;
+EXPOSE_SINGLE_INTERFACE_GLOBALVAR( CSoundSystem, ISoundSystem, SOUNDSYSTEM_INTERFACE_VERSION, s_SoundSystem );
+
+
+//-----------------------------------------------------------------------------
+// Connect, disconnect
+//-----------------------------------------------------------------------------
+bool CSoundSystem::Connect( CreateInterfaceFn factory )
+{
+	if ( !BaseClass::Connect( factory ) )
+		return false;
+
+	g_pDataCache = (IDataCache*)factory( DATACACHE_INTERFACE_VERSION, NULL );
+	g_pSoundSystem = this;
+	return (g_pFullFileSystem != NULL) && (g_pDataCache != NULL);
+}
+
+void CSoundSystem::Disconnect()
+{
+	g_pSoundSystem = NULL;
+	g_pDataCache = NULL;
+	BaseClass::Disconnect();
+}
+
+
+//-----------------------------------------------------------------------------
+// Query interface
+//-----------------------------------------------------------------------------
+void *CSoundSystem::QueryInterface( const char *pInterfaceName )
+{
+	if (!Q_strncmp(	pInterfaceName, SOUNDSYSTEM_INTERFACE_VERSION, Q_strlen(SOUNDSYSTEM_INTERFACE_VERSION) + 1))
+		return (ISoundSystem*)this;
+
+	return NULL;
+}
+
+
+//-----------------------------------------------------------------------------
+// Init, shutdown
+//-----------------------------------------------------------------------------
+InitReturnVal_t CSoundSystem::Init()
+{
+	InitReturnVal_t nRetVal = BaseClass::Init();
+	if ( nRetVal != INIT_OK )
+		return nRetVal;
+
+	MathLib_Init( 2.2f, 2.2f, 0.0f, 2.0f );
+	m_flElapsedTime = 0.0f;
+	m_pAudioDevice = Audio_CreateWaveDevice();
+	if ( !m_pAudioDevice->Init() )
+		return INIT_FAILED;
+
+	return INIT_OK;
+}
+
+void CSoundSystem::Shutdown()
+{
+	Msg( "Removing %i sounds\n", m_ActiveSounds.Size() );
+	for ( int i = 0 ; i < m_ActiveSounds.Size(); i++ )
+	{
+		CSoundFile *p = &m_ActiveSounds[ i ];
+		Msg( "Removing sound:  %s\n", p->filename );
+		delete p->source;
+	}
+
+	m_ActiveSounds.RemoveAll();
+
+	if ( m_pAudioDevice )
+	{
+		m_pAudioDevice->Shutdown();
+		delete m_pAudioDevice;
+	}
+
+	BaseClass::Shutdown();
+}
+
+
+//-----------------------------------------------------------------------------
+// Purpose: 
+//-----------------------------------------------------------------------------
+CAudioSource *CSoundSystem::FindOrAddSound( const char *filename )
+{
+	CSoundFile *s;
+
+	int i;
+	for ( i = 0; i < m_ActiveSounds.Size(); i++ )
+	{
+		s = &m_ActiveSounds[ i ];
+		Assert( s );
+		if ( !stricmp( s->filename, filename ) )
+		{
+			long filetime = g_pFullFileSystem->GetFileTime( filename );
+			if ( filetime != s->filetime )
+			{
+				Msg( "Reloading sound %s\n", filename );
+				delete s->source;
+				s->source = LoadSound( filename );
+				s->filetime = filetime;
+			}
+			return s->source;
+		}
+	}
+
+	i = m_ActiveSounds.AddToTail();
+	s = &m_ActiveSounds[ i ];
+	strcpy( s->filename, filename );
+	s->source = LoadSound( filename );
+	s->filetime = g_pFullFileSystem->GetFileTime( filename );
+
+	return s->source;
+}
+
+CAudioSource *CSoundSystem::LoadSound( const char *wavfile )
+{
+	if ( !m_pAudioDevice )
+		return NULL;
+
+	CAudioSource *wave = AudioSource_Create( wavfile );
+	return wave;
+}
+
+void CSoundSystem::PlaySound( CAudioSource *source, float volume, CAudioMixer **ppMixer )
+{
+	if ( ppMixer )
+	{
+		*ppMixer = NULL;
+	}
+
+	if ( m_pAudioDevice )
+	{
+		CAudioMixer *mixer = source->CreateMixer();
+		if ( ppMixer )
+		{
+			*ppMixer = mixer;
+		}
+		mixer->SetVolume( volume );
+		m_pAudioDevice->AddSource( mixer );
+	}
+}
+
+void CSoundSystem::Update( float dt )
+{
+//	closecaptionmanager->PreProcess( g_nSoundFrameCount );
+
+	if ( m_pAudioDevice )
+	{
+		m_pAudioDevice->Update( m_flElapsedTime );
+	}
+
+//	closecaptionmanager->PostProcess( g_nSoundFrameCount, dt );
+
+	m_flElapsedTime += dt;
+	g_nSoundFrameCount++;
+}
+
+void CSoundSystem::Flush( void )
+{
+	if ( m_pAudioDevice )
+	{
+		m_pAudioDevice->Flush();
+	}
+}
+
+void CSoundSystem::StopAll( void )
+{
+	if ( m_pAudioDevice )
+	{
+		m_pAudioDevice->StopSounds();
+	}
+}
+
+void CSoundSystem::StopSound( CAudioMixer *mixer )
+{
+	int idx = m_pAudioDevice->FindSourceIndex( mixer );
+	if ( idx != -1 )
+	{
+		m_pAudioDevice->FreeChannel( idx );
+	}
+}
+
+bool CSoundSystem::IsSoundPlaying( CAudioMixer *pMixer )
+{
+	if ( !m_pAudioDevice || !pMixer )
+		return false;
+
+	//
+	int index = m_pAudioDevice->FindSourceIndex( pMixer );
+	if ( index != -1 )
+		return true;
+
+	return false;
+}
+
+CAudioMixer *CSoundSystem::FindMixer( CAudioSource *source )
+{
+	if ( !m_pAudioDevice )
+		return NULL;
+
+	return m_pAudioDevice->GetMixerForSource( source );
+}

soundsystem/soundsystem.h

@@ -0,0 +1,35 @@
+//========= Copyright Valve Corporation, All rights reserved. ============//
+//
+// Purpose: DLL interface for low-level sound utilities
+//
+//===========================================================================//
+
+#ifndef SOUNDSYSTEM_H
+#define SOUNDSYSTEM_H
+
+#ifdef _WIN32
+#pragma once
+#endif
+
+
+#include "tier2/tier2.h"
+#include "tier3/tier3.h"
+
+
+//-----------------------------------------------------------------------------
+// Forward declarations
+//-----------------------------------------------------------------------------
+class ISoundSystem;
+class ISoundSystemServices;
+class IAudioDevice;
+
+
+//-----------------------------------------------------------------------------
+// Singleton interface
+//-----------------------------------------------------------------------------
+extern ISoundSystem *g_pSoundSystem;
+extern ISoundSystemServices *g_pSoundServices;
+extern IAudioDevice *g_pAudioDevice;
+
+
+#endif // SOUNDSYSTEM_H

soundsystem/soundsystem.vpc

@@ -0,0 +1,64 @@
+//-----------------------------------------------------------------------------
+//	SOUNDSYSTEM.VPC
+//
+//	Project Script
+//-----------------------------------------------------------------------------
+
+$Macro SRCDIR		".."
+$Macro OUTBINDIR	"$SRCDIR\..\game\bin"
+
+$Include "$SRCDIR\vpc_scripts\source_dll_base.vpc"
+
+$Configuration
+{
+	$Compiler
+	{
+		$PreprocessorDefinitions			"$BASE;SOUNDSYSTEM_EXPORTS"
+	}
+
+	$Linker
+	{
+		$AdditionalDependencies				"$BASE winmm.lib"
+	}
+}
+
+$Project "Soundsystem"
+{
+	$Folder	"Source Files"
+	{
+		$File	"$SRCDIR\public\sentence.cpp"
+		$File	"snd_audio_source.cpp"
+		$File	"snd_dev_wave.cpp"
+		$File	"snd_io.cpp"
+		$File	"snd_wave_mixer.cpp"
+		$File	"snd_wave_mixer_adpcm.cpp"
+		$File	"snd_wave_source.cpp"
+		$File	"soundsystem.cpp"
+	}
+
+	$Folder	"Header Files"
+	{
+		$File	"snd_dev_wave.h"
+		$File	"snd_wave_mixer.h"
+		$File	"snd_wave_mixer_adpcm.h"
+		$File	"snd_wave_mixer_private.h"
+		$File	"snd_wave_source.h"
+		$File	"soundsystem.h"
+	}
+
+	$Folder	"Interface"
+	{
+		$File	"$SRCDIR\public\soundsystem\isoundsystem.h"
+		$File	"$SRCDIR\public\soundsystem\snd_audio_source.h"
+		$File	"$SRCDIR\public\soundsystem\snd_device.h"
+	}
+
+	$Folder	"Link Libraries"
+	{
+		$Lib mathlib
+		$Lib tier2
+		$File	"$SRCDIR\dx9sdk\lib\dsound.lib"
+		$File	"$SRCDIR\dx9sdk\lib\dxguid.lib"
+		$File	"$SRCDIR\lib\common\mss32.lib"
+	}
+}