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1782 lines
40 KiB
1782 lines
40 KiB
//===== Copyright © 1996-2005, Valve Corporation, All rights reserved. ======//
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//
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// Purpose:
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//
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// $NoKeywords: $
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//
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//===========================================================================//
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#define WIN32_LEAN_AND_MEAN
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#include <windows.h>
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#pragma warning( disable : 4201 )
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#include <mmsystem.h>
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#include <stdio.h>
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#include <math.h>
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#include "color.h"
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#include "snd_audio_source.h"
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#include "AudioWaveOutput.h"
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#include "ifaceposersound.h"
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#include "StudioModel.h"
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#include "hlfaceposer.h"
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#include "expressions.h"
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#include "expclass.h"
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#include "PhonemeConverter.h"
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#include "utlvector.h"
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#include "FileSystem.h"
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#include "sentence.h"
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#include "faceposer_models.h"
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#include "iclosecaptionmanager.h"
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#include "phonemeeditor.h"
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#include "wavebrowser.h"
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#include "choreoscene.h"
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#include "choreoview.h"
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typedef struct channel_s
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{
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int leftvol;
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int rightvol;
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int rleftvol;
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int rrightvol;
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float pitch;
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} channel_t;
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#define INPUT_BUFFER_COUNT 32
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class CAudioWaveInput : public CAudioInput
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{
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public:
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CAudioWaveInput( void );
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~CAudioWaveInput( void );
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// Returns the current count of available samples
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int SampleCount( void );
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// returns the size of each sample in bytes
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int SampleSize( void ) { return m_sampleSize; }
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// returns the sampling rate of the data
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int SampleRate( void ) { return m_sampleRate; }
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// returns a pointer to the actual data
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void *SampleData( void );
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// release the available data (mark as done)
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void SampleRelease( void );
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// returns the mono/stereo status of this device (true if stereo)
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bool IsStereo( void ) { return m_isStereo; }
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// begin sampling
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void Start( void );
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// stop sampling
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void Stop( void );
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void WaveMessage( HWAVEIN hdevice, UINT uMsg, DWORD dwParam1, DWORD dwParam2 );
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private:
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void OpenDevice( void );
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bool ValidDevice( void ) { return m_deviceId != 0xFFFFFFFF; }
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void ClearDevice( void ) { m_deviceId = 0xFFFFFFFF; }
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// returns true if the new format is better
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bool BetterFormat( DWORD dwNewFormat, DWORD dwOldFormat );
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void InitReadyList( void );
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void AddToReadyList( WAVEHDR *pBuffer );
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void PopReadyList( void );
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WAVEHDR *m_pReadyList;
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int m_sampleSize;
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int m_sampleRate;
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bool m_isStereo;
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UINT m_deviceId;
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HWAVEIN m_deviceHandle;
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WAVEHDR *m_buffers[ INPUT_BUFFER_COUNT ];
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};
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extern "C" void CALLBACK WaveData( HWAVEIN hwi, UINT uMsg, CAudioWaveInput *pAudio, DWORD dwParam1, DWORD dwParam2 );
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CAudioWaveInput::CAudioWaveInput( void )
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{
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memset( m_buffers, 0, sizeof( m_buffers ) );
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int deviceCount = (int)waveInGetNumDevs();
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UINT deviceId = 0;
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DWORD deviceFormat = 0;
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int i;
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for ( i = 0; i < deviceCount; i++ )
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{
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WAVEINCAPS waveCaps;
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MMRESULT errorCode = waveInGetDevCaps( (UINT)i, &waveCaps, sizeof(waveCaps) );
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if ( errorCode == MMSYSERR_NOERROR )
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{
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// valid device
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if ( BetterFormat( waveCaps.dwFormats, deviceFormat ) )
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{
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deviceId = i;
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deviceFormat = waveCaps.dwFormats;
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}
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}
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}
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if ( !deviceFormat )
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{
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m_deviceId = 0xFFFFFFFF;
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m_sampleSize = 0;
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m_sampleRate = 0;
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m_isStereo = false;
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}
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else
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{
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m_deviceId = deviceId;
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m_sampleRate = 44100;
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m_isStereo = false;
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if ( deviceFormat & WAVE_FORMAT_4M16 )
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{
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m_sampleSize = 2;
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}
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else if ( deviceFormat & WAVE_FORMAT_4M08 )
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{
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m_sampleSize = 1;
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}
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else
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{
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// ERROR!
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}
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OpenDevice();
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}
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InitReadyList();
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}
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CAudioWaveInput::~CAudioWaveInput( void )
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{
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if ( ValidDevice() )
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{
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Stop();
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waveInReset( m_deviceHandle );
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waveInClose( m_deviceHandle );
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for ( int i = 0; i < INPUT_BUFFER_COUNT; i++ )
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{
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if ( m_buffers[i] )
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{
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waveInUnprepareHeader( m_deviceHandle, m_buffers[i], sizeof( *m_buffers[i] ) );
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delete[] m_buffers[i]->lpData;
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delete m_buffers[i];
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}
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m_buffers[i] = NULL;
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}
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ClearDevice();
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}
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}
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void CALLBACK WaveData( HWAVEIN hwi, UINT uMsg, CAudioWaveInput *pAudio, DWORD dwParam1, DWORD dwParam2 )
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{
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if ( pAudio )
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{
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pAudio->WaveMessage( hwi, uMsg, dwParam1, dwParam2 );
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}
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}
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void CAudioWaveInput::WaveMessage( HWAVEIN hdevice, UINT uMsg, DWORD dwParam1, DWORD dwParam2 )
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{
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if ( hdevice != m_deviceHandle )
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return;
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switch( uMsg )
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{
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case WIM_DATA:
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WAVEHDR *pHeader = (WAVEHDR *)dwParam1;
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AddToReadyList( pHeader );
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break;
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}
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}
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void CAudioWaveInput::OpenDevice( void )
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{
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if ( !ValidDevice() )
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return;
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WAVEFORMATEX format;
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memset( &format, 0, sizeof(format) );
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format.nAvgBytesPerSec = m_sampleRate * m_sampleSize;
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format.nChannels = 1;
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format.wBitsPerSample = m_sampleSize * 8;
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format.nSamplesPerSec = m_sampleRate;
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format.wFormatTag = WAVE_FORMAT_PCM;
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format.nBlockAlign = m_sampleSize;
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MMRESULT errorCode = waveInOpen( &m_deviceHandle, m_deviceId, &format, (DWORD)WaveData, (DWORD)this, CALLBACK_FUNCTION );
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if ( errorCode == MMSYSERR_NOERROR )
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{
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// valid device opened
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int bufferSize = m_sampleSize * m_sampleRate / INPUT_BUFFER_COUNT; // total of one second of data
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// allocate buffers
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for ( int i = 0; i < INPUT_BUFFER_COUNT; i++ )
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{
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m_buffers[i] = new WAVEHDR;
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m_buffers[i]->lpData = new char[ bufferSize ];
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m_buffers[i]->dwBufferLength = bufferSize;
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m_buffers[i]->dwUser = 0;
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m_buffers[i]->dwFlags = 0;
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waveInPrepareHeader( m_deviceHandle, m_buffers[i], sizeof( *m_buffers[i] ) );
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waveInAddBuffer( m_deviceHandle, m_buffers[i], sizeof( *m_buffers[i] ) );
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}
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}
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else
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{
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ClearDevice();
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}
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}
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void CAudioWaveInput::Start( void )
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{
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if ( !ValidDevice() )
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return;
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waveInStart( m_deviceHandle );
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}
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void CAudioWaveInput::Stop( void )
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{
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if ( !ValidDevice() )
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return;
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waveInStop( m_deviceHandle );
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}
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void CAudioWaveInput::InitReadyList( void )
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{
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m_pReadyList = NULL;
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}
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void CAudioWaveInput::AddToReadyList( WAVEHDR *pBuffer )
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{
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WAVEHDR **pList = &m_pReadyList;
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waveInUnprepareHeader( m_deviceHandle, pBuffer, sizeof(*pBuffer) );
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// insert at the tail of the list
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while ( *pList )
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{
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pList = reinterpret_cast<WAVEHDR **>(&((*pList)->dwUser));
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}
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pBuffer->dwUser = NULL;
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*pList = pBuffer;
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}
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void CAudioWaveInput::PopReadyList( void )
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{
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if ( m_pReadyList )
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{
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WAVEHDR *pBuffer = m_pReadyList;
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m_pReadyList = reinterpret_cast<WAVEHDR *>(m_pReadyList->dwUser);
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waveInPrepareHeader( m_deviceHandle, pBuffer, sizeof(*pBuffer) );
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waveInAddBuffer( m_deviceHandle, pBuffer, sizeof(*pBuffer) );
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}
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}
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#define WAVE_FORMAT_STEREO (WAVE_FORMAT_1S08|WAVE_FORMAT_1S16|WAVE_FORMAT_2S08|WAVE_FORMAT_2S16|WAVE_FORMAT_4S08|WAVE_FORMAT_4S16)
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#define WAVE_FORMATS_UNDERSTOOD (0xFFF)
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#define WAVE_FORMAT_11K (WAVE_FORMAT_1M08|WAVE_FORMAT_1M16)
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#define WAVE_FORMAT_22K (WAVE_FORMAT_2M08|WAVE_FORMAT_2M16)
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#define WAVE_FORMAT_44K (WAVE_FORMAT_4M08|WAVE_FORMAT_4M16)
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static int HighestBit( DWORD dwFlags )
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{
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int i = 31;
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while ( i )
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{
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if ( dwFlags & (1<<i) )
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return i;
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i--;
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}
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return 0;
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}
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bool CAudioWaveInput::BetterFormat( DWORD dwNewFormat, DWORD dwOldFormat )
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{
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dwNewFormat &= WAVE_FORMATS_UNDERSTOOD & (~WAVE_FORMAT_STEREO);
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dwOldFormat &= WAVE_FORMATS_UNDERSTOOD & (~WAVE_FORMAT_STEREO);
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// our target format is 44.1KHz, mono, 16-bit
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if ( HighestBit(dwOldFormat) >= HighestBit(dwNewFormat) )
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return false;
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return true;
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}
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int CAudioWaveInput::SampleCount( void )
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{
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if ( !ValidDevice() )
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return 0;
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if ( m_pReadyList )
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{
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switch( SampleSize() )
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{
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case 2:
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return m_pReadyList->dwBytesRecorded >> 1;
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case 1:
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return m_pReadyList->dwBytesRecorded;
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default:
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break;
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}
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}
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return 0;
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}
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void *CAudioWaveInput::SampleData( void )
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{
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if ( !ValidDevice() )
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return NULL;
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if ( m_pReadyList )
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{
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return m_pReadyList->lpData;
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}
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return NULL;
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}
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// release the available data (mark as done)
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void CAudioWaveInput::SampleRelease( void )
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{
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PopReadyList();
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}
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// factory to create a suitable audio input for this system
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CAudioInput *CAudioInput::Create( void )
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{
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// sound source is a singleton for now
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static CAudioInput *pSource = NULL;
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if ( !pSource )
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{
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pSource = new CAudioWaveInput;
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}
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return pSource;
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}
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void CAudioDeviceSWMix::Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, int rateScaleFix, int outCount, int timecompress, bool forward )
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{
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int sampleIndex = 0;
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fixedint sampleFrac = inputOffset;
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int fixup = 0;
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int fixupstep = 1;
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if ( !forward )
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{
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fixup = outCount - 1;
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fixupstep = -1;
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}
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for ( int i = 0; i < outCount; i++, fixup += fixupstep )
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{
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int dest = max( outputOffset + fixup, 0 );
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m_paintbuffer[ dest ].left += pChannel->leftvol * pData[sampleIndex];
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m_paintbuffer[ dest ].right += pChannel->rightvol * pData[sampleIndex];
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sampleFrac += rateScaleFix;
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sampleIndex += FIX_INTPART(sampleFrac);
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sampleFrac = FIX_FRACPART(sampleFrac);
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}
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}
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void CAudioDeviceSWMix::Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, int rateScaleFix, int outCount, int timecompress, bool forward )
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{
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int sampleIndex = 0;
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fixedint sampleFrac = inputOffset;
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int fixup = 0;
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int fixupstep = 1;
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if ( !forward )
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{
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fixup = outCount - 1;
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fixupstep = -1;
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}
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for ( int i = 0; i < outCount; i++, fixup += fixupstep )
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{
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int dest = max( outputOffset + fixup, 0 );
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m_paintbuffer[ dest ].left += pChannel->leftvol * pData[sampleIndex];
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m_paintbuffer[ dest ].right += pChannel->rightvol * pData[sampleIndex+1];
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sampleFrac += rateScaleFix;
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sampleIndex += FIX_INTPART(sampleFrac)<<1;
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sampleFrac = FIX_FRACPART(sampleFrac);
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}
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}
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void CAudioDeviceSWMix::Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, int rateScaleFix, int outCount, int timecompress, bool forward )
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{
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int sampleIndex = 0;
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fixedint sampleFrac = inputOffset;
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int fixup = 0;
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int fixupstep = 1;
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if ( !forward )
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{
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fixup = outCount - 1;
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fixupstep = -1;
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}
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for ( int i = 0; i < outCount; i++, fixup += fixupstep )
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{
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int dest = max( outputOffset + fixup, 0 );
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m_paintbuffer[ dest ].left += (pChannel->leftvol * pData[sampleIndex])>>8;
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m_paintbuffer[ dest ].right += (pChannel->rightvol * pData[sampleIndex])>>8;
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sampleFrac += rateScaleFix;
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sampleIndex += FIX_INTPART(sampleFrac);
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sampleFrac = FIX_FRACPART(sampleFrac);
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}
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}
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void CAudioDeviceSWMix::Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, int rateScaleFix, int outCount, int timecompress, bool forward )
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{
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int sampleIndex = 0;
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fixedint sampleFrac = inputOffset;
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int fixup = 0;
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int fixupstep = 1;
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if ( !forward )
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{
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fixup = outCount - 1;
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fixupstep = -1;
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}
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for ( int i = 0; i < outCount; i++, fixup += fixupstep )
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{
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int dest = max( outputOffset + fixup, 0 );
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m_paintbuffer[ dest ].left += (pChannel->leftvol * pData[sampleIndex])>>8;
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m_paintbuffer[ dest ].right += (pChannel->rightvol * pData[sampleIndex+1])>>8;
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sampleFrac += rateScaleFix;
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sampleIndex += FIX_INTPART(sampleFrac)<<1;
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sampleFrac = FIX_FRACPART(sampleFrac);
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}
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}
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int CAudioDeviceSWMix::MaxSampleCount( void )
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{
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return PAINTBUFFER_SIZE;
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}
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void CAudioDeviceSWMix::MixBegin( void )
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{
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memset( m_paintbuffer, 0, sizeof(m_paintbuffer) );
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}
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void CAudioDeviceSWMix::TransferBufferStereo16( short *pOutput, int sampleCount )
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{
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for ( int i = 0; i < sampleCount; i++ )
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{
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if ( m_paintbuffer[i].left > 32767 )
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m_paintbuffer[i].left = 32767;
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else if ( m_paintbuffer[i].left < -32768 )
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m_paintbuffer[i].left = -32768;
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if ( m_paintbuffer[i].right > 32767 )
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m_paintbuffer[i].right = 32767;
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else if ( m_paintbuffer[i].right < -32768 )
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m_paintbuffer[i].right = -32768;
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*pOutput++ = (short)m_paintbuffer[i].left;
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*pOutput++ = (short)m_paintbuffer[i].right;
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}
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}
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CAudioWaveOutput::CAudioWaveOutput( void )
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{
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for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
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{
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CAudioBuffer *buffer = &m_buffers[ i ];
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Assert( buffer );
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buffer->hdr = NULL;
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buffer->submitted = false;
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buffer->submit_sample_count = false;
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}
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ClearDevice();
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OpenDevice();
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m_mixTime = -1;
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m_sampleIndex = 0;
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memset( m_sourceList, 0, sizeof(m_sourceList) );
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m_nEstimatedSamplesAhead = (int)( ( float ) OUTPUT_SAMPLE_RATE / 10.0f );
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}
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void CAudioWaveOutput::RemoveMixerChannelReferences( CAudioMixer *mixer )
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{
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for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
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{
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RemoveFromReferencedList( mixer, &m_buffers[ i ] );
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}
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}
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void CAudioWaveOutput::AddToReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer )
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{
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// Already in list
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for ( int i = 0; i < buffer->m_Referenced.Count(); i++ )
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{
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if ( buffer->m_Referenced[ i ].mixer == mixer )
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{
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return;
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}
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}
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|
|
// Just remove it
|
|
int idx = buffer->m_Referenced.AddToTail();
|
|
|
|
CAudioMixerState *state = &buffer->m_Referenced[ idx ];
|
|
state->mixer = mixer;
|
|
state->submit_mixer_sample = mixer->GetSamplePosition();
|
|
|
|
}
|
|
|
|
void CAudioWaveOutput::RemoveFromReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer )
|
|
{
|
|
for ( int i = 0; i < buffer->m_Referenced.Count(); i++ )
|
|
{
|
|
if ( buffer->m_Referenced[ i ].mixer == mixer )
|
|
{
|
|
buffer->m_Referenced.Remove( i );
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
bool CAudioWaveOutput::IsSoundInReferencedList( CAudioMixer *mixer, CAudioBuffer *buffer )
|
|
{
|
|
for ( int i = 0; i < buffer->m_Referenced.Count(); i++ )
|
|
{
|
|
if ( buffer->m_Referenced[ i ].mixer == mixer )
|
|
{
|
|
return true;
|
|
}
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool CAudioWaveOutput::IsSourceReferencedByActiveBuffer( CAudioMixer *mixer )
|
|
{
|
|
if ( !ValidDevice() )
|
|
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;
|
|
}
|
|
|
|
CAudioWaveOutput::~CAudioWaveOutput( void )
|
|
{
|
|
if ( ValidDevice() )
|
|
{
|
|
waveOutReset( m_deviceHandle );
|
|
for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
|
|
{
|
|
if ( m_buffers[i].hdr )
|
|
{
|
|
waveOutUnprepareHeader( m_deviceHandle, m_buffers[i].hdr, sizeof(*m_buffers[i].hdr) );
|
|
delete[] m_buffers[i].hdr->lpData;
|
|
delete m_buffers[i].hdr;
|
|
}
|
|
m_buffers[i].hdr = NULL;
|
|
m_buffers[i].submitted = false;
|
|
m_buffers[i].submit_sample_count = 0;
|
|
m_buffers[i].m_Referenced.Purge();
|
|
}
|
|
waveOutClose( m_deviceHandle );
|
|
ClearDevice();
|
|
}
|
|
}
|
|
|
|
|
|
|
|
CAudioBuffer *CAudioWaveOutput::GetEmptyBuffer( void )
|
|
{
|
|
CAudioBuffer *pOutput = NULL;
|
|
if ( ValidDevice() )
|
|
{
|
|
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 CAudioWaveOutput::SilenceBuffer( short *pSamples, int sampleCount )
|
|
{
|
|
int i;
|
|
|
|
for ( i = 0; i < sampleCount; i++ )
|
|
{
|
|
// left
|
|
*pSamples++ = 0;
|
|
// right
|
|
*pSamples++ = 0;
|
|
}
|
|
}
|
|
|
|
void CAudioWaveOutput::Flush( void )
|
|
{
|
|
waveOutReset( m_deviceHandle );
|
|
}
|
|
|
|
// mix a buffer up to time (time is absolute)
|
|
void CAudioWaveOutput::Update( float time )
|
|
{
|
|
if ( !ValidDevice() )
|
|
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 > m_audioDevice.MaxSampleCount() )
|
|
sampleCount = m_audioDevice.MaxSampleCount();
|
|
else
|
|
sampleCount = tempCount;
|
|
|
|
m_audioDevice.MixBegin();
|
|
for ( int i = 0; i < MAX_CHANNELS; i++ )
|
|
{
|
|
CAudioMixer *pSource = m_sourceList[i];
|
|
if ( !pSource )
|
|
continue;
|
|
|
|
StudioModel *model = NULL;
|
|
|
|
int modelindex = pSource->GetModelIndex();
|
|
if ( modelindex >= 0 )
|
|
{
|
|
model = models->GetStudioModel( modelindex );
|
|
}
|
|
else
|
|
{
|
|
if ( g_pPhonemeEditor->IsActiveTool() || g_pWaveBrowser->IsActiveTool() )
|
|
{
|
|
model = models->GetActiveStudioModel();
|
|
|
|
}
|
|
}
|
|
|
|
if ( model && !model->m_mouth.IsSourceReferenced( pSource->GetSource() ) )
|
|
{
|
|
CChoreoScene *pScene = g_pChoreoView->GetScene();
|
|
bool bIgnorePhonemes = pScene ? pScene->ShouldIgnorePhonemes() : false;
|
|
model->m_mouth.AddSource( pSource->GetSource(), bIgnorePhonemes );
|
|
if ( modelindex < 0 )
|
|
{
|
|
pSource->SetModelIndex( models->GetIndexForStudioModel( model ) );
|
|
}
|
|
}
|
|
|
|
int currentsample = pSource->GetSamplePosition();
|
|
bool forward = pSource->GetDirection();
|
|
|
|
if ( pSource->GetActive() )
|
|
{
|
|
if ( !pSource->MixDataToDevice( &m_audioDevice, pSource->GetChannel(), currentsample, sampleCount, SampleRate(), 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, SampleRate(), forward );
|
|
}
|
|
}
|
|
|
|
}
|
|
|
|
m_audioDevice.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_deviceHandle, pBuffer->hdr, sizeof(*(pBuffer->hdr)) );
|
|
}
|
|
}
|
|
|
|
int CAudioWaveOutput::GetNumberofSamplesAhead( void )
|
|
{
|
|
ComputeSampleAheadAmount();
|
|
return m_nEstimatedSamplesAhead;
|
|
}
|
|
|
|
float CAudioWaveOutput::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 CAudioWaveOutput::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 CAudioWaveOutput::FindSourceIndex( CAudioMixer *pSource )
|
|
{
|
|
for ( int i = 0; i < MAX_CHANNELS; i++ )
|
|
{
|
|
if ( pSource == m_sourceList[i] )
|
|
{
|
|
return i;
|
|
}
|
|
}
|
|
return -1;
|
|
}
|
|
|
|
CAudioMixer *CAudioWaveOutput::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 CAudioWaveOutput::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 CAudioWaveOutput::StopSounds( void )
|
|
{
|
|
for ( int i = 0; i < MAX_CHANNELS; i++ )
|
|
{
|
|
if ( m_sourceList[i] )
|
|
{
|
|
FreeChannel( i );
|
|
}
|
|
}
|
|
}
|
|
|
|
|
|
void CAudioWaveOutput::SetChannel( int channelIndex, CAudioMixer *pSource )
|
|
{
|
|
if ( channelIndex < 0 || channelIndex >= MAX_CHANNELS )
|
|
return;
|
|
|
|
m_sourceList[channelIndex] = pSource;
|
|
}
|
|
|
|
void CAudioWaveOutput::FreeChannel( int channelIndex )
|
|
{
|
|
if ( channelIndex < 0 || channelIndex >= MAX_CHANNELS )
|
|
return;
|
|
|
|
if ( m_sourceList[channelIndex] )
|
|
{
|
|
StudioModel *model = NULL;
|
|
int modelindex = m_sourceList[channelIndex]->GetModelIndex();
|
|
if ( modelindex >= 0)
|
|
{
|
|
model = models->GetStudioModel( modelindex );
|
|
}
|
|
|
|
if ( model )
|
|
{
|
|
model->m_mouth.RemoveSource( m_sourceList[channelIndex]->GetSource() );
|
|
}
|
|
|
|
RemoveMixerChannelReferences( m_sourceList[channelIndex] );
|
|
|
|
delete m_sourceList[channelIndex];
|
|
m_sourceList[channelIndex] = NULL;
|
|
}
|
|
}
|
|
|
|
int CAudioWaveOutput::GetOutputPosition( void )
|
|
{
|
|
if ( !m_deviceHandle )
|
|
return 0;
|
|
|
|
MMTIME mmtime;
|
|
mmtime.wType = TIME_SAMPLES;
|
|
waveOutGetPosition( m_deviceHandle, &mmtime, sizeof( MMTIME ) );
|
|
|
|
// Convert time to sample count
|
|
return ( mmtime.u.sample );
|
|
}
|
|
|
|
void CAudioWaveOutput::OpenDevice( void )
|
|
{
|
|
WAVEFORMATEX waveFormat;
|
|
|
|
memset( &waveFormat, 0, sizeof(waveFormat) );
|
|
// Select a PCM, 16-bit stereo playback device
|
|
waveFormat.cbSize = sizeof(waveFormat);
|
|
waveFormat.nAvgBytesPerSec = OUTPUT_SAMPLE_RATE * 2 * 2;
|
|
waveFormat.nBlockAlign = 2 * 2; // channels * sample size
|
|
waveFormat.nChannels = 2; // stereo
|
|
waveFormat.nSamplesPerSec = OUTPUT_SAMPLE_RATE;
|
|
waveFormat.wBitsPerSample = 16;
|
|
waveFormat.wFormatTag = WAVE_FORMAT_PCM;
|
|
|
|
MMRESULT errorCode = waveOutOpen( &m_deviceHandle, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL );
|
|
if ( errorCode == MMSYSERR_NOERROR )
|
|
{
|
|
int bufferSize = 4 * ( OUTPUT_SAMPLE_RATE / OUTPUT_BUFFER_COUNT ); // total of 1 second of data
|
|
|
|
// Got one!
|
|
for ( int i = 0; i < OUTPUT_BUFFER_COUNT; i++ )
|
|
{
|
|
m_buffers[i].hdr = new WAVEHDR;
|
|
m_buffers[i].hdr->lpData = new char[ bufferSize ];
|
|
long align = (long)m_buffers[i].hdr->lpData;
|
|
if ( align & 3 )
|
|
{
|
|
m_buffers[i].hdr->lpData = (char *) ( (align+3) &~3 );
|
|
}
|
|
m_buffers[i].hdr->dwBufferLength = bufferSize - (align&3);
|
|
m_buffers[i].hdr->dwFlags = 0;
|
|
|
|
if ( waveOutPrepareHeader( m_deviceHandle, m_buffers[i].hdr, sizeof(*m_buffers[i].hdr) ) != MMSYSERR_NOERROR )
|
|
{
|
|
ClearDevice();
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
else
|
|
{
|
|
ClearDevice();
|
|
}
|
|
}
|
|
|
|
// factory to create a suitable audio output for this system
|
|
CAudioOutput *CAudioOutput::Create( void )
|
|
{
|
|
// sound device is a singleton for now
|
|
static CAudioOutput *pWaveOut = NULL;
|
|
|
|
if ( !pWaveOut )
|
|
{
|
|
pWaveOut = new CAudioWaveOutput;
|
|
}
|
|
|
|
return pWaveOut;
|
|
}
|
|
|
|
struct CSoundFile
|
|
{
|
|
char filename[ 512 ];
|
|
CAudioSource *source;
|
|
long filetime;
|
|
};
|
|
|
|
//-----------------------------------------------------------------------------
|
|
// Purpose:
|
|
//-----------------------------------------------------------------------------
|
|
class CFacePoserSound : public IFacePoserSound
|
|
{
|
|
public:
|
|
~CFacePoserSound( void );
|
|
|
|
void Init( void );
|
|
void Shutdown( void );
|
|
void Update( float dt );
|
|
void Flush( void );
|
|
|
|
CAudioSource *LoadSound( const char *wavfile );
|
|
void PlaySound( StudioModel *source, float volume, const char *wavfile, CAudioMixer **ppMixer );
|
|
void PlaySound( CAudioSource *source, float volume, CAudioMixer **ppMixer );
|
|
void PlayPartialSound( StudioModel *model, float volume, const char *wavfile, CAudioMixer **ppMixer, int startSample, int endSample );
|
|
|
|
bool IsSoundPlaying( CAudioMixer *pMixer );
|
|
CAudioMixer *FindMixer( CAudioSource *source );
|
|
|
|
void StopAll( void );
|
|
void StopSound( CAudioMixer *mixer );
|
|
|
|
void RenderWavToDC( HDC dc, RECT& outrect, const Color& clr, float starttime, float endtime,
|
|
CAudioSource *pWave, bool selected = false, int selectionstart = 0, int selectionend = 0 );
|
|
|
|
// void InstallPhonemecallback( IPhonemeTag *pTagInterface );
|
|
float GetAmountofTimeAhead( void );
|
|
|
|
int GetNumberofSamplesAhead( void );
|
|
|
|
CAudioOuput *GetAudioOutput( void );
|
|
|
|
virtual void EnsureNoModelReferences( CAudioSource *source );
|
|
|
|
private:
|
|
void AddViseme( float intensity, StudioModel *model, int phoneme, float scale );
|
|
void ProcessCloseCaptionData( StudioModel *model, float curtime, CSentence* sentence );
|
|
void SetupWeights( void );
|
|
|
|
CAudioSource *FindOrAddSound( const char *filename );
|
|
|
|
CAudioOutput *m_pAudio;
|
|
|
|
float m_flElapsedTime;
|
|
|
|
CUtlVector < CSoundFile > m_ActiveSounds;
|
|
};
|
|
|
|
static CFacePoserSound g_FacePoserSound;
|
|
|
|
IFacePoserSound *sound = ( IFacePoserSound * )&g_FacePoserSound;
|
|
|
|
CFacePoserSound::~CFacePoserSound( void )
|
|
{
|
|
OutputDebugString( va( "Removing %i sounds\n", m_ActiveSounds.Count() ) );
|
|
for ( int i = 0 ; i < m_ActiveSounds.Count(); i++ )
|
|
{
|
|
CSoundFile *p = &m_ActiveSounds[ i ];
|
|
OutputDebugString( va( "Removing sound: %s\n", p->filename ) );
|
|
delete p->source;
|
|
}
|
|
|
|
m_ActiveSounds.RemoveAll();
|
|
}
|
|
|
|
//-----------------------------------------------------------------------------
|
|
// Purpose:
|
|
//-----------------------------------------------------------------------------
|
|
CAudioOuput *CFacePoserSound::GetAudioOutput( void )
|
|
{
|
|
return (CAudioOuput *)m_pAudio;
|
|
}
|
|
|
|
CAudioSource *CFacePoserSound::FindOrAddSound( const char *filename )
|
|
{
|
|
CSoundFile *s;
|
|
|
|
int i;
|
|
for ( i = 0; i < m_ActiveSounds.Count(); i++ )
|
|
{
|
|
s = &m_ActiveSounds[ i ];
|
|
Assert( s );
|
|
if ( !stricmp( s->filename, filename ) )
|
|
{
|
|
long filetime = filesystem->GetFileTime( filename );
|
|
if ( filetime != s->filetime )
|
|
{
|
|
Con_Printf( "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 = filesystem->GetFileTime( filename );
|
|
|
|
return s->source;
|
|
}
|
|
|
|
void CFacePoserSound::Init( void )
|
|
{
|
|
m_flElapsedTime = 0.0f;
|
|
m_pAudio = CAudioOutput::Create();
|
|
}
|
|
|
|
void CFacePoserSound::Shutdown( void )
|
|
{
|
|
}
|
|
|
|
float CFacePoserSound::GetAmountofTimeAhead( void )
|
|
{
|
|
if ( !m_pAudio )
|
|
return 0.0f;
|
|
|
|
return m_pAudio->GetAmountofTimeAhead();
|
|
}
|
|
|
|
int CFacePoserSound::GetNumberofSamplesAhead( void )
|
|
{
|
|
if ( !m_pAudio )
|
|
return 0;
|
|
|
|
return m_pAudio->GetNumberofSamplesAhead();
|
|
}
|
|
|
|
|
|
CAudioSource *CFacePoserSound::LoadSound( const char *wavfile )
|
|
{
|
|
if ( !m_pAudio )
|
|
return NULL;
|
|
|
|
CAudioSource *wave = AudioSource_Create( wavfile );
|
|
return wave;
|
|
}
|
|
|
|
void CFacePoserSound::PlaySound( StudioModel *model, float volume, const char *wavfile, CAudioMixer **ppMixer )
|
|
{
|
|
if ( m_pAudio )
|
|
{
|
|
CAudioSource *wave = FindOrAddSound( wavfile );
|
|
if ( !wave )
|
|
return;
|
|
|
|
CAudioMixer *pMixer = wave->CreateMixer();
|
|
if ( ppMixer )
|
|
{
|
|
*ppMixer = pMixer;
|
|
}
|
|
pMixer->SetVolume( volume );
|
|
m_pAudio->AddSource( pMixer );
|
|
if ( model )
|
|
{
|
|
pMixer->SetModelIndex( models->GetIndexForStudioModel( model ) );
|
|
}
|
|
}
|
|
}
|
|
|
|
void CFacePoserSound::PlayPartialSound( StudioModel *model, float volume, const char *wavfile, CAudioMixer **ppMixer, int startSample, int endSample )
|
|
{
|
|
if ( !m_pAudio )
|
|
return;
|
|
|
|
StopAll();
|
|
CAudioSource *wave = FindOrAddSound( wavfile );
|
|
if ( !wave )
|
|
return;
|
|
|
|
CAudioMixer *mixer = wave->CreateMixer();
|
|
if ( ppMixer )
|
|
{
|
|
*ppMixer = mixer;
|
|
}
|
|
|
|
mixer->SetSamplePosition( startSample );
|
|
mixer->SetLoopPosition( endSample );
|
|
mixer->SetVolume( volume );
|
|
m_pAudio->AddSource( mixer );
|
|
}
|
|
|
|
void CFacePoserSound::PlaySound( CAudioSource *source, float volume, CAudioMixer **ppMixer )
|
|
{
|
|
if ( ppMixer )
|
|
{
|
|
*ppMixer = NULL;
|
|
}
|
|
|
|
if ( m_pAudio )
|
|
{
|
|
CAudioMixer *mixer = source->CreateMixer();
|
|
if ( ppMixer )
|
|
{
|
|
*ppMixer = mixer;
|
|
}
|
|
mixer->SetVolume( volume );
|
|
m_pAudio->AddSource( mixer );
|
|
}
|
|
}
|
|
|
|
enum
|
|
{
|
|
PHONEME_CLASS_WEAK = 0,
|
|
PHONEME_CLASS_NORMAL,
|
|
PHONEME_CLASS_STRONG,
|
|
|
|
NUM_PHONEME_CLASSES
|
|
};
|
|
|
|
struct Emphasized_Phoneme
|
|
{
|
|
char *classname;
|
|
bool required;
|
|
bool valid;
|
|
CExpClass *cl;
|
|
CExpression *exp;
|
|
float *settings;
|
|
float amount;
|
|
};
|
|
|
|
static Emphasized_Phoneme g_PhonemeClasses[ NUM_PHONEME_CLASSES ] =
|
|
{
|
|
{ "phonemes_weak", false },
|
|
{ "phonemes", true },
|
|
{ "phonemes_strong", false },
|
|
};
|
|
|
|
#define STRONG_CROSSFADE_START 0.60f
|
|
#define WEAK_CROSSFADE_START 0.40f
|
|
|
|
void ComputeBlendedSetting( Emphasized_Phoneme *classes, float emphasis_intensity )
|
|
{
|
|
// Here's the formula
|
|
// 0.5 is neutral 100 % of the default setting
|
|
|
|
// Crossfade starts at STRONG_CROSSFADE_START and is full at STRONG_CROSSFADE_END
|
|
// If there isn't a strong then the intensity of the underlying phoneme is fixed at 2 x STRONG_CROSSFADE_START
|
|
// so we don't get huge numbers
|
|
|
|
bool has_weak = classes[ PHONEME_CLASS_WEAK ].valid;
|
|
bool has_strong = classes[ PHONEME_CLASS_STRONG ].valid;
|
|
|
|
Assert( classes[ PHONEME_CLASS_NORMAL ].valid );
|
|
|
|
if ( emphasis_intensity > STRONG_CROSSFADE_START )
|
|
{
|
|
if ( has_strong )
|
|
{
|
|
// Blend in some of strong
|
|
float dist_remaining = 1.0f - emphasis_intensity;
|
|
float frac = dist_remaining / ( 1.0f - STRONG_CROSSFADE_START );
|
|
|
|
classes[ PHONEME_CLASS_NORMAL ].amount = (frac) * 2.0f * STRONG_CROSSFADE_START;
|
|
classes[ PHONEME_CLASS_STRONG ].amount = 1.0f - frac;
|
|
}
|
|
else
|
|
{
|
|
emphasis_intensity = min( emphasis_intensity, STRONG_CROSSFADE_START );
|
|
classes[ PHONEME_CLASS_NORMAL ].amount = 2.0f * emphasis_intensity;
|
|
}
|
|
}
|
|
else if ( emphasis_intensity < WEAK_CROSSFADE_START )
|
|
{
|
|
if ( has_weak )
|
|
{
|
|
// Blend in some weak
|
|
float dist_remaining = WEAK_CROSSFADE_START - emphasis_intensity;
|
|
float frac = dist_remaining / ( WEAK_CROSSFADE_START );
|
|
|
|
classes[ PHONEME_CLASS_NORMAL ].amount = (1.0f - frac) * 2.0f * WEAK_CROSSFADE_START;
|
|
classes[ PHONEME_CLASS_WEAK ].amount = frac;
|
|
}
|
|
else
|
|
{
|
|
emphasis_intensity = max( emphasis_intensity, WEAK_CROSSFADE_START );
|
|
classes[ PHONEME_CLASS_NORMAL ].amount = 2.0f * emphasis_intensity;
|
|
}
|
|
}
|
|
else
|
|
{
|
|
classes[ PHONEME_CLASS_NORMAL ].amount = 2.0f * emphasis_intensity;
|
|
}
|
|
}
|
|
|
|
void CFacePoserSound::AddViseme( float intensity, StudioModel *model, int phoneme, float scale )
|
|
{
|
|
int i;
|
|
|
|
Assert( model );
|
|
CStudioHdr *hdr = model->GetStudioHdr();
|
|
Assert( hdr );
|
|
if ( !hdr )
|
|
return;
|
|
|
|
for ( i = 0; i < NUM_PHONEME_CLASSES; i++ )
|
|
{
|
|
Emphasized_Phoneme *info = &g_PhonemeClasses[ i ];
|
|
|
|
info->valid = false;
|
|
info->exp = NULL;
|
|
info->settings = NULL;
|
|
info->amount = 0.0f;
|
|
|
|
info->cl = expressions->FindClass( info->classname, true );
|
|
if ( info->cl )
|
|
{
|
|
info->exp = info->cl->FindExpression( ConvertPhoneme( phoneme ) );
|
|
}
|
|
|
|
if ( info->required && ( !info->cl || !info->exp ) )
|
|
{
|
|
return;
|
|
}
|
|
|
|
if ( info->exp )
|
|
{
|
|
info->valid = true;
|
|
|
|
info->settings = info->exp->GetSettings();
|
|
Assert( info->settings );
|
|
}
|
|
}
|
|
|
|
ComputeBlendedSetting( g_PhonemeClasses, intensity );
|
|
|
|
// Look up the phoneme
|
|
for ( LocalFlexController_t i = LocalFlexController_t(0); i < hdr->numflexcontrollers(); i++)
|
|
{
|
|
int j = hdr->pFlexcontroller( i )->localToGlobal;
|
|
|
|
float add = 0.0f;
|
|
|
|
for ( int k = 0 ; k < NUM_PHONEME_CLASSES; k++ )
|
|
{
|
|
Emphasized_Phoneme *info = &g_PhonemeClasses[ k ];
|
|
if ( !info->valid || !info->amount )
|
|
continue;
|
|
|
|
add += info->amount * info->settings[ j ];
|
|
}
|
|
|
|
if ( add == 0.0f )
|
|
continue;
|
|
|
|
float curvalue = model->GetFlexController( i );
|
|
curvalue += add * scale;
|
|
model->SetFlexController( i, curvalue );
|
|
}
|
|
}
|
|
|
|
|
|
#define PHONEME_FILTER 0.08f
|
|
#define PHONEME_DELAY 0.0f
|
|
|
|
void CFacePoserSound::SetupWeights( void )
|
|
{
|
|
StudioModel *model;
|
|
int c = models->Count();
|
|
for ( int i = 0; i < c; i++ )
|
|
{
|
|
model = models->GetStudioModel( i );
|
|
if ( !model )
|
|
continue;
|
|
|
|
// Reset flexes
|
|
CStudioHdr *hdr = model->GetStudioHdr();
|
|
if ( !hdr )
|
|
continue;
|
|
|
|
for ( int s = 0; s < model->m_mouth.GetNumVoiceSources(); s++ )
|
|
{
|
|
CVoiceData *vd = model->m_mouth.GetVoiceSource( s );
|
|
if ( !vd || vd->ShouldIgnorePhonemes() )
|
|
continue;
|
|
|
|
CAudioSource *source = vd->GetSource();
|
|
// check for phoneme flexes
|
|
if ( !source )
|
|
continue;
|
|
|
|
CAudioMixer *mixer = FindMixer( source );
|
|
if ( !mixer )
|
|
continue;
|
|
|
|
CSentence *sentence = source->GetSentence();
|
|
if ( !sentence )
|
|
continue;
|
|
|
|
// Zero faces if needed
|
|
models->CheckResetFlexes();
|
|
|
|
float pos = (float)mixer->GetScrubPosition();
|
|
|
|
// Con_Printf( "pos %f for mixer %p\n", pos, mixer );
|
|
|
|
float soundtime = pos / source->SampleRate();
|
|
|
|
float t = soundtime - PHONEME_DELAY;
|
|
float dt = PHONEME_FILTER;
|
|
|
|
float sentence_duration = source->GetRunningLength();
|
|
float emphasis_intensity = sentence->GetIntensity( t, sentence_duration );
|
|
|
|
if ( t > 0.0f )
|
|
{
|
|
for ( int w = 0 ; w < sentence->m_Words.Count(); w++ )
|
|
{
|
|
CWordTag *word = sentence->m_Words[ w ];
|
|
if ( !word )
|
|
continue;
|
|
|
|
for ( int k = 0; k < word->m_Phonemes.Count(); k++)
|
|
{
|
|
CPhonemeTag *phoneme = word->m_Phonemes[ k ];
|
|
if ( !phoneme )
|
|
continue;
|
|
|
|
// if the filter starts within this phoneme, make sure the filter size is
|
|
// at least least as long as the current phoneme, or until the end of the next phoneme,
|
|
// whichever is smaller
|
|
if (t > phoneme->GetStartTime() && t < phoneme->GetEndTime())
|
|
{
|
|
CPhonemeTag *next = NULL;
|
|
// try next phoneme, or first phoneme of next word
|
|
if (k < word->m_Phonemes.Count()-1)
|
|
{
|
|
next = word->m_Phonemes[ k+1 ];
|
|
}
|
|
else if ( w < sentence->m_Words.Count() - 1 && sentence->m_Words[ w+1 ]->m_Phonemes.Count() )
|
|
{
|
|
next = sentence->m_Words[ w+1 ]->m_Phonemes[ 0 ];
|
|
}
|
|
|
|
// if I have a neighbor
|
|
if (next)
|
|
{
|
|
// and they're touching
|
|
if (next->GetStartTime() == phoneme->GetEndTime())
|
|
{
|
|
// no gap, so increase the blend length to the end of the next phoneme, as long as it's not longer than the current phoneme
|
|
dt = max( dt, min( next->GetEndTime() - t, phoneme->GetEndTime() - phoneme->GetStartTime() ) );
|
|
}
|
|
else
|
|
{
|
|
// dead space, so increase the blend length to the start of the next phoneme, as long as it's not longer than the current phoneme
|
|
dt = max( dt, min( next->GetStartTime() - t, phoneme->GetEndTime() - phoneme->GetStartTime() ) );
|
|
}
|
|
}
|
|
else
|
|
{
|
|
// last phoneme in list, increase the blend length to the length of the current phoneme
|
|
dt = max( dt, phoneme->GetEndTime() - phoneme->GetStartTime() );
|
|
}
|
|
}
|
|
|
|
float t1 = ( phoneme->GetStartTime() - t) / dt;
|
|
float t2 = ( phoneme->GetEndTime() - t) / dt;
|
|
|
|
if (t1 < 1.0 && t2 > 0)
|
|
{
|
|
float scale;
|
|
|
|
// clamp
|
|
if (t2 > 1)
|
|
t2 = 1;
|
|
if (t1 < 0)
|
|
t1 = 0;
|
|
|
|
// FIXME: simple box filter. Should use something fancier
|
|
scale = (t2 - t1);
|
|
|
|
AddViseme( emphasis_intensity, model, phoneme->GetPhonemeCode(), scale );
|
|
}
|
|
}
|
|
}
|
|
ProcessCloseCaptionData( model, t, sentence );
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
static int g_nSoundFrameCount = 0;
|
|
|
|
void CFacePoserSound::ProcessCloseCaptionData( StudioModel *model, float curtime, CSentence* sentence )
|
|
{
|
|
// closecaptionmanager->Process( g_nSoundFrameCount, model, curtime, sentence, GetCloseCaptionLanguageId() );
|
|
}
|
|
|
|
void CFacePoserSound::Update( float dt )
|
|
{
|
|
// closecaptionmanager->PreProcess( g_nSoundFrameCount );
|
|
|
|
if ( m_pAudio )
|
|
{
|
|
SetupWeights();
|
|
m_pAudio->Update( m_flElapsedTime );
|
|
}
|
|
|
|
// closecaptionmanager->PostProcess( g_nSoundFrameCount, dt );
|
|
|
|
m_flElapsedTime += dt;
|
|
g_nSoundFrameCount++;
|
|
}
|
|
|
|
void CFacePoserSound::Flush( void )
|
|
{
|
|
if ( m_pAudio )
|
|
{
|
|
m_pAudio->Flush();
|
|
}
|
|
}
|
|
|
|
void CFacePoserSound::StopAll( void )
|
|
{
|
|
int c = models->Count();
|
|
for ( int i = 0; i < c; i++ )
|
|
{
|
|
StudioModel *model = models->GetStudioModel( i );
|
|
if ( model )
|
|
{
|
|
model->m_mouth.ClearVoiceSources();
|
|
}
|
|
}
|
|
|
|
if ( m_pAudio )
|
|
{
|
|
m_pAudio->StopSounds();
|
|
}
|
|
}
|
|
|
|
void CFacePoserSound::StopSound( CAudioMixer *mixer )
|
|
{
|
|
int idx = m_pAudio->FindSourceIndex( mixer );
|
|
if ( idx != -1 )
|
|
{
|
|
m_pAudio->FreeChannel( idx );
|
|
}
|
|
}
|
|
|
|
void CFacePoserSound::RenderWavToDC( HDC dc, RECT& outrect, const Color& clr,
|
|
float starttime, float endtime, CAudioSource *pWave,
|
|
bool selected /*= false*/, int selectionstart /*= 0*/, int selectionend /*= 0*/ )
|
|
{
|
|
channel_t channel;
|
|
|
|
channel.leftvol = 127;
|
|
channel.rightvol = 127;
|
|
channel.pitch = 1.0;
|
|
|
|
if ( !pWave )
|
|
return;
|
|
|
|
CAudioWaveOutput *pWaveOutput = ( CAudioWaveOutput * )m_pAudio;
|
|
|
|
CAudioMixer *pMixer = pWave->CreateMixer();
|
|
|
|
float timeperpixel = ( endtime - starttime ) / (float)( outrect.right - outrect.left );
|
|
|
|
float samplesperpixel = timeperpixel * pWave->SampleRate();
|
|
|
|
samplesperpixel = min( samplesperpixel, (float)PAINTBUFFER_SIZE );
|
|
|
|
int intsamplesperpixel = (int)samplesperpixel;
|
|
|
|
// Determine start/stop positions
|
|
int totalsamples = (int)( pWave->GetRunningLength() * pWave->SampleRate() );
|
|
|
|
if ( totalsamples <= 0 )
|
|
return;
|
|
|
|
float selectionstarttime = pWave->GetRunningLength() * ( float )selectionstart / ( float )totalsamples;
|
|
float selectionendtime = pWave->GetRunningLength() * ( float )selectionend / ( float )totalsamples;
|
|
|
|
|
|
HPEN oldPen, pen, pen2, pen3, pen4;
|
|
|
|
pen = CreatePen( PS_SOLID, 1, ColorToRGB( Color( 175, 175, 250 ) ) );
|
|
pen2 = CreatePen( PS_SOLID, 1, ColorToRGB( clr ) );
|
|
pen3 = CreatePen( PS_SOLID, 1, ColorToRGB( Color( 127, 200, 249 ) ) );
|
|
pen4 = CreatePen( PS_SOLID, 2, ColorToRGB( Color( 0, 0, 200 ) ) );
|
|
|
|
oldPen = (HPEN)SelectObject( dc, pen );
|
|
|
|
MoveToEx( dc, outrect.left, ( outrect.bottom + outrect.top ) / 2, NULL );
|
|
LineTo( dc, outrect.right, ( outrect.bottom + outrect.top ) / 2 );
|
|
|
|
SelectObject( dc, pen2 );
|
|
|
|
// Now iterate the samples
|
|
float currenttime = 0.0f;
|
|
int pixel = 0;
|
|
int height = ( outrect.bottom - outrect.top ) / 2;
|
|
int midy = ( outrect.bottom + outrect.top ) / 2;
|
|
int bufferlen = ( intsamplesperpixel + 3 ) & ~3;
|
|
short *samples = new short[ 2 * bufferlen ];
|
|
bool drawingselection = false;
|
|
int maxsamples = max( 32, intsamplesperpixel / 16 );
|
|
int currentsample = 0;
|
|
|
|
while ( currenttime < endtime )
|
|
{
|
|
|
|
pWaveOutput->m_audioDevice.MixBegin();
|
|
|
|
int samplecount = min( maxsamples, intsamplesperpixel );
|
|
|
|
if ( !pMixer->MixDataToDevice( &pWaveOutput->m_audioDevice, &channel, currentsample, samplecount, pWave->SampleRate(), true ) )
|
|
break;
|
|
|
|
currentsample = pMixer->GetSamplePosition();
|
|
|
|
// Jump ahead by diff
|
|
int diff = intsamplesperpixel - samplecount;
|
|
if ( diff > 0 )
|
|
{
|
|
if ( !pMixer->SkipSamples( &channel, currentsample, diff, pWave->SampleRate(), true ) )
|
|
break;
|
|
}
|
|
|
|
currentsample = pMixer->GetSamplePosition();
|
|
|
|
pWaveOutput->m_audioDevice.TransferBufferStereo16( samples, samplecount );
|
|
|
|
if ( currenttime >= starttime )
|
|
{
|
|
if ( selected )
|
|
{
|
|
bool boundary = false;
|
|
bool inselection = ( currenttime >= selectionstarttime &&
|
|
currenttime <= selectionendtime );
|
|
|
|
if ( inselection )
|
|
{
|
|
if ( !drawingselection )
|
|
{
|
|
drawingselection = true;
|
|
boundary = true;
|
|
}
|
|
}
|
|
else if ( drawingselection )
|
|
{
|
|
boundary = true;
|
|
drawingselection = false;
|
|
}
|
|
|
|
if ( inselection || boundary )
|
|
{
|
|
int top, bottom;
|
|
|
|
bottom = outrect.bottom;
|
|
|
|
HPEN *usePen;
|
|
if ( boundary )
|
|
{
|
|
usePen = &pen4;
|
|
top = outrect.top;
|
|
}
|
|
else
|
|
{
|
|
usePen = &pen3;
|
|
top = outrect.bottom - 19;
|
|
}
|
|
|
|
HPEN old = (HPEN)SelectObject( dc, *usePen );
|
|
|
|
MoveToEx( dc, outrect.left + pixel, top, NULL );
|
|
LineTo( dc, outrect.left + pixel, bottom-1 );
|
|
|
|
SelectObject( dc, old );
|
|
}
|
|
}
|
|
|
|
|
|
int maxvalue = -65536;
|
|
int minvalue = 65536;
|
|
|
|
short *pData = samples;
|
|
|
|
// only take fix samples
|
|
int step = 2;
|
|
int count = 2 * samplecount;
|
|
|
|
for ( int i = 0; i < count; i+=step )
|
|
{
|
|
int val = (int)( pData[i] + pData[i+1] ) / 2;
|
|
|
|
if ( val > maxvalue )
|
|
{
|
|
maxvalue = val;
|
|
}
|
|
|
|
if ( val < minvalue )
|
|
{
|
|
minvalue = val;
|
|
}
|
|
}
|
|
|
|
float maxv = (float)( maxvalue ) / 32768.0f;
|
|
float minv = (float)( minvalue ) / 32768.0f;
|
|
|
|
MoveToEx( dc, outrect.left + pixel, midy + ( int ) ( maxv * height ), NULL );
|
|
LineTo( dc, outrect.left + pixel, midy + ( int ) ( minv * height ) );
|
|
|
|
pixel++;
|
|
}
|
|
currenttime += timeperpixel;
|
|
}
|
|
|
|
delete[] samples;
|
|
|
|
SelectObject( dc, oldPen );
|
|
DeleteObject( pen );
|
|
DeleteObject( pen2 );
|
|
DeleteObject( pen3 );
|
|
|
|
delete pMixer;
|
|
}
|
|
|
|
bool CFacePoserSound::IsSoundPlaying( CAudioMixer *pMixer )
|
|
{
|
|
if ( !m_pAudio || !pMixer )
|
|
{
|
|
return false;
|
|
}
|
|
|
|
//
|
|
int index = m_pAudio->FindSourceIndex( pMixer );
|
|
if ( index != -1 )
|
|
return true;
|
|
|
|
return false;
|
|
}
|
|
|
|
CAudioMixer *CFacePoserSound::FindMixer( CAudioSource *source )
|
|
{
|
|
if ( !m_pAudio )
|
|
return NULL;
|
|
|
|
return m_pAudio->GetMixerForSource( source );
|
|
}
|
|
|
|
|
|
void CFacePoserSound::EnsureNoModelReferences( CAudioSource *source )
|
|
{
|
|
int c = models->Count();
|
|
for ( int i = 0; i < c; i++ )
|
|
{
|
|
StudioModel *model = models->GetStudioModel( i );
|
|
if ( model->m_mouth.IsSourceReferenced( source ) )
|
|
{
|
|
model->m_mouth.RemoveSource( source );
|
|
}
|
|
}
|
|
}
|