612 lines
16 KiB
C++
612 lines
16 KiB
C++
//========= Copyright Valve Corporation, All rights reserved. ============//
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//
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// Purpose:
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//
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//===========================================================================//
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#include "audio_pch.h"
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#include <OpenAL/al.h>
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#include <OpenAL/alc.h>
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#ifdef OSX
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#include <OpenAL/MacOSX_OALExtensions.h>
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#endif
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// memdbgon must be the last include file in a .cpp file!!!
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#include "tier0/memdbgon.h"
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#ifndef DEDICATED // have to test this because VPC is forcing us to compile this file.
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extern bool snd_firsttime;
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extern bool MIX_ScaleChannelVolume( paintbuffer_t *ppaint, channel_t *pChannel, int volume[CCHANVOLUMES], int mixchans );
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extern void S_SpatializeChannel( int volume[6], int master_vol, const Vector *psourceDir, float gain, float mono );
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#define NUM_BUFFERS_SOURCES 128
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#define BUFF_MASK (NUM_BUFFERS_SOURCES - 1 )
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#define BUFFER_SIZE 0x0400
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//-----------------------------------------------------------------------------
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//
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// NOTE: This only allows 16-bit, stereo wave out
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//
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//-----------------------------------------------------------------------------
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class CAudioDeviceOpenAL : public CAudioDeviceBase
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{
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public:
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bool IsActive( void );
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bool Init( void );
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void Shutdown( void );
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void PaintEnd( void );
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int GetOutputPosition( void );
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void ChannelReset( int entnum, int channelIndex, float distanceMod );
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void Pause( void );
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void UnPause( void );
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float MixDryVolume( void );
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bool Should3DMix( void );
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void StopAllSounds( void );
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int PaintBegin( float mixAheadTime, int soundtime, int paintedtime );
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void ClearBuffer( void );
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void UpdateListener( const Vector& position, const Vector& forward, const Vector& right, const Vector& up );
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void MixBegin( int sampleCount );
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void MixUpsample( int sampleCount, int filtertype );
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void Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress );
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void TransferSamples( int end );
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void SpatializeChannel( int volume[CCHANVOLUMES/2], int master_vol, const Vector& sourceDir, float gain, float mono);
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void ApplyDSPEffects( int idsp, portable_samplepair_t *pbuffront, portable_samplepair_t *pbufrear, portable_samplepair_t *pbufcenter, int samplecount );
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const char *DeviceName( void ) { return "OpenAL"; }
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int DeviceChannels( void ) { return 2; }
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int DeviceSampleBits( void ) { return 16; }
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int DeviceSampleBytes( void ) { return 2; }
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int DeviceDmaSpeed( void ) { return SOUND_DMA_SPEED; }
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int DeviceSampleCount( void ) { return m_deviceSampleCount; }
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private:
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void OpenWaveOut( void );
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void CloseWaveOut( void );
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bool ValidWaveOut( void ) const;
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ALuint m_Buffer[NUM_BUFFERS_SOURCES];
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ALuint m_Source[1];
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int m_SndBufSize;
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void *m_sndBuffers;
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int m_deviceSampleCount;
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int m_buffersSent;
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int m_buffersCompleted;
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int m_pauseCount;
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bool m_bSoundsShutdown;
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};
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IAudioDevice *Audio_CreateOpenALDevice( void )
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{
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CAudioDeviceOpenAL *wave = NULL;
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if ( !wave )
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{
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wave = new CAudioDeviceOpenAL;
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}
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if ( wave->Init() )
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return wave;
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delete wave;
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wave = NULL;
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return NULL;
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}
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void OnSndSurroundCvarChanged( IConVar *pVar, const char *pOldString, float flOldValue );
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void OnSndSurroundLegacyChanged( IConVar *pVar, const char *pOldString, float flOldValue );
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//-----------------------------------------------------------------------------
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// Init, shutdown
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//-----------------------------------------------------------------------------
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bool CAudioDeviceOpenAL::Init( void )
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{
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m_SndBufSize = 0;
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m_sndBuffers = NULL;
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m_pauseCount = 0;
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m_bSurround = false;
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m_bSurroundCenter = false;
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m_bHeadphone = false;
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m_buffersSent = 0;
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m_buffersCompleted = 0;
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m_pauseCount = 0;
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m_bSoundsShutdown = false;
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static bool first = true;
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if ( first )
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{
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snd_surround.SetValue( 2 );
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snd_surround.InstallChangeCallback( &OnSndSurroundCvarChanged );
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snd_legacy_surround.InstallChangeCallback( &OnSndSurroundLegacyChanged );
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first = false;
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}
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OpenWaveOut();
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if ( snd_firsttime )
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{
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DevMsg( "Wave sound initialized\n" );
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}
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return ValidWaveOut();
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}
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void CAudioDeviceOpenAL::Shutdown( void )
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{
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CloseWaveOut();
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}
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//-----------------------------------------------------------------------------
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// WAV out device
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//-----------------------------------------------------------------------------
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inline bool CAudioDeviceOpenAL::ValidWaveOut( void ) const
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{
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return m_sndBuffers != 0;
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}
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//-----------------------------------------------------------------------------
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// Opens the windows wave out device
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//-----------------------------------------------------------------------------
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void CAudioDeviceOpenAL::OpenWaveOut( void )
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{
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m_buffersSent = 0;
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m_buffersCompleted = 0;
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ALenum error;
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ALCcontext *newContext = NULL;
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ALCdevice *newDevice = NULL;
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// Create a new OpenAL Device
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// Pass NULL to specify the system‚use default output device
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const ALCchar *initStr = (const ALCchar *)"\'( (sampling-rate 44100 ))";
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newDevice = alcOpenDevice(initStr);
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if (newDevice != NULL)
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{
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// Create a new OpenAL Context
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// The new context will render to the OpenAL Device just created
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ALCint attr[] = { ALC_FREQUENCY, DeviceDmaSpeed(), ALC_SYNC, AL_FALSE, 0 };
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newContext = alcCreateContext(newDevice, attr );
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if (newContext != NULL)
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{
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// Make the new context the Current OpenAL Context
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alcMakeContextCurrent(newContext);
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// Create some OpenAL Buffer Objects
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alGenBuffers( NUM_BUFFERS_SOURCES, m_Buffer);
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if((error = alGetError()) != AL_NO_ERROR)
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{
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DevMsg("Error Generating Buffers: ");
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return;
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}
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// Create some OpenAL Source Objects
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alGenSources(1, m_Source);
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if(alGetError() != AL_NO_ERROR)
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{
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DevMsg("Error generating sources! \n");
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return;
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}
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alListener3f( AL_POSITION,0.0f,0.0f,0.0f);
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int i;
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for ( i = 0; i < 1; i++ )
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{
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alSource3f( m_Source[i],AL_POSITION,0.0f,0.0f,0.0f );
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alSourcef( m_Source[i], AL_PITCH, 1.0f );
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alSourcef( m_Source[i], AL_GAIN, 1.0f );
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}
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}
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}
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m_SndBufSize = NUM_BUFFERS_SOURCES*BUFFER_SIZE;
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m_deviceSampleCount = m_SndBufSize / DeviceSampleBytes();
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if ( !m_sndBuffers )
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{
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m_sndBuffers = malloc( m_SndBufSize );
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memset( m_sndBuffers, 0x0, m_SndBufSize );
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}
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}
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//-----------------------------------------------------------------------------
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// Closes the windows wave out device
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//-----------------------------------------------------------------------------
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void CAudioDeviceOpenAL::CloseWaveOut( void )
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{
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if ( ValidWaveOut() )
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{
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ALCcontext *context = NULL;
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ALCdevice *device = NULL;
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m_bSoundsShutdown = true;
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alSourceStop( m_Source[0] );
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// Delete the Sources
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alDeleteSources(1, m_Source);
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// Delete the Buffers
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alDeleteBuffers(NUM_BUFFERS_SOURCES, m_Buffer);
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//Get active context
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context = alcGetCurrentContext();
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//Get device for active context
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device = alcGetContextsDevice(context);
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alcMakeContextCurrent( NULL );
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alcSuspendContext(context);
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//Release context
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alcDestroyContext(context);
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//Close device
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alcCloseDevice(device);
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}
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if ( m_sndBuffers )
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{
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free( m_sndBuffers );
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m_sndBuffers = NULL;
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}
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}
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//-----------------------------------------------------------------------------
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// Mixing setup
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//-----------------------------------------------------------------------------
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int CAudioDeviceOpenAL::PaintBegin( float mixAheadTime, int soundtime, int paintedtime )
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{
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// soundtime - total samples that have been played out to hardware at dmaspeed
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// paintedtime - total samples that have been mixed at speed
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// endtime - target for samples in mixahead buffer at speed
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unsigned int endtime = soundtime + mixAheadTime * DeviceDmaSpeed();
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int samps = DeviceSampleCount() >> (DeviceChannels()-1);
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if ((int)(endtime - soundtime) > samps)
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endtime = soundtime + samps;
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if ((endtime - paintedtime) & 0x3)
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{
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// The difference between endtime and painted time should align on
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// boundaries of 4 samples. This is important when upsampling from 11khz -> 44khz.
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endtime -= (endtime - paintedtime) & 0x3;
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}
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return endtime;
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}
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#ifdef OSX
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ALvoid alBufferDataStaticProc(const ALint bid, ALenum format, ALvoid* data, ALsizei size, ALsizei freq)
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{
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static alBufferDataStaticProcPtr proc = NULL;
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if (proc == NULL) {
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proc = (alBufferDataStaticProcPtr) alGetProcAddress((const ALCchar*) "alBufferDataStatic");
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}
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if (proc)
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proc(bid, format, data, size, freq);
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}
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#endif
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//-----------------------------------------------------------------------------
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// Actually performs the mixing
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//-----------------------------------------------------------------------------
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void CAudioDeviceOpenAL::PaintEnd( void )
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{
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if ( !m_sndBuffers /*|| m_bSoundsShutdown*/ )
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return;
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ALint state;
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ALenum error;
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int iloop;
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int cblocks = 4 << 1;
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ALint processed = 1;
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ALuint lastUnqueuedBuffer = 0;
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ALuint unqueuedBuffer = -1;
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int nProcessedLoop = 200; // spin for a max of 200 times de-queing buffers, fixes a hang on exit
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while ( processed > 0 && --nProcessedLoop > 0 )
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{
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alGetSourcei( m_Source[ 0 ], AL_BUFFERS_PROCESSED, &processed);
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error = alGetError();
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if (error != AL_NO_ERROR)
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break;
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if ( processed > 0 )
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{
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lastUnqueuedBuffer = unqueuedBuffer;
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alSourceUnqueueBuffers( m_Source[ 0 ], 1, &unqueuedBuffer );
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error = alGetError();
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if ( error != AL_NO_ERROR && error != AL_INVALID_NAME )
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{
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DevMsg( "Error alSourceUnqueueBuffers %d\n", error );
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break;
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}
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else
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{
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m_buffersCompleted++; // this buffer has been played
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}
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}
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}
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//
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// submit a few new sound blocks
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//
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// 44K sound support
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while (((m_buffersSent - m_buffersCompleted) >> SAMPLE_16BIT_SHIFT) < cblocks)
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{
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int iBuf = m_buffersSent&BUFF_MASK;
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#ifdef OSX
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alBufferDataStaticProc( m_Buffer[iBuf], AL_FORMAT_STEREO16, (char *)m_sndBuffers + iBuf*BUFFER_SIZE, BUFFER_SIZE, DeviceDmaSpeed() );
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#else
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alBufferData( m_Buffer[iBuf], AL_FORMAT_STEREO16, (char *)m_sndBuffers + iBuf*BUFFER_SIZE, BUFFER_SIZE, DeviceDmaSpeed() );
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#endif
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if ( (error = alGetError()) != AL_NO_ERROR )
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{
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DevMsg( "Error alBufferData %d %d\n", iBuf, error );
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}
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alSourceQueueBuffers( m_Source[0], 1, &m_Buffer[iBuf] );
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if ( (error = alGetError() ) != AL_NO_ERROR )
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{
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DevMsg( "Error alSourceQueueBuffers %d %d\n", iBuf, error );
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}
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m_buffersSent++;
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}
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// make sure the stream is playing
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alGetSourcei( m_Source[ 0 ], AL_SOURCE_STATE, &state);
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if ( state != AL_PLAYING )
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{
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DevMsg( "Restarting sound playback\n" );
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alSourcePlay( m_Source[0] );
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if((error = alGetError()) != AL_NO_ERROR)
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{
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DevMsg( "Error alSourcePlay %d\n", error );
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}
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}
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}
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int CAudioDeviceOpenAL::GetOutputPosition( void )
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{
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int s = m_buffersSent * BUFFER_SIZE;
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s >>= SAMPLE_16BIT_SHIFT;
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s &= (DeviceSampleCount()-1);
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return s / DeviceChannels();
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}
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//-----------------------------------------------------------------------------
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// Pausing
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//-----------------------------------------------------------------------------
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void CAudioDeviceOpenAL::Pause( void )
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{
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m_pauseCount++;
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if (m_pauseCount == 1)
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{
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alSourceStop( m_Source[0] );
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}
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}
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void CAudioDeviceOpenAL::UnPause( void )
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{
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if ( m_pauseCount > 0 )
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{
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m_pauseCount--;
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}
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if ( m_pauseCount == 0 )
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{
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alSourcePlay( m_Source[0] );
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}
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}
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bool CAudioDeviceOpenAL::IsActive( void )
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{
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return ( m_pauseCount == 0 );
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}
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float CAudioDeviceOpenAL::MixDryVolume( void )
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{
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return 0;
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}
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bool CAudioDeviceOpenAL::Should3DMix( void )
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{
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return false;
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}
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void CAudioDeviceOpenAL::ClearBuffer( void )
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{
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if ( !m_sndBuffers )
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return;
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Q_memset( m_sndBuffers, 0x0, DeviceSampleCount() * DeviceSampleBytes() );
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}
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void CAudioDeviceOpenAL::UpdateListener( const Vector& position, const Vector& forward, const Vector& right, const Vector& up )
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{
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}
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void CAudioDeviceOpenAL::MixBegin( int sampleCount )
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{
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MIX_ClearAllPaintBuffers( sampleCount, false );
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}
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void CAudioDeviceOpenAL::MixUpsample( int sampleCount, int filtertype )
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{
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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int ifilter = ppaint->ifilter;
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Assert (ifilter < CPAINTFILTERS);
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S_MixBufferUpsample2x( sampleCount, ppaint->pbuf, &(ppaint->fltmem[ifilter][0]), CPAINTFILTERMEM, filtertype );
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ppaint->ifilter++;
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}
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void CAudioDeviceOpenAL::Mix8Mono( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 1))
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return;
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Mix8MonoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, (byte *)pData, inputOffset, rateScaleFix, outCount );
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}
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void CAudioDeviceOpenAL::Mix8Stereo( channel_t *pChannel, char *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 2 ))
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return;
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Mix8StereoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, (byte *)pData, inputOffset, rateScaleFix, outCount );
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}
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void CAudioDeviceOpenAL::Mix16Mono( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 1 ))
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return;
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Mix16MonoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, pData, inputOffset, rateScaleFix, outCount );
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}
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void CAudioDeviceOpenAL::Mix16Stereo( channel_t *pChannel, short *pData, int outputOffset, int inputOffset, fixedint rateScaleFix, int outCount, int timecompress )
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{
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int volume[CCHANVOLUMES];
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paintbuffer_t *ppaint = MIX_GetCurrentPaintbufferPtr();
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if (!MIX_ScaleChannelVolume( ppaint, pChannel, volume, 2 ))
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return;
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Mix16StereoWavtype( pChannel, ppaint->pbuf + outputOffset, volume, pData, inputOffset, rateScaleFix, outCount );
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}
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void CAudioDeviceOpenAL::ChannelReset( int entnum, int channelIndex, float distanceMod )
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{
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}
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void CAudioDeviceOpenAL::TransferSamples( int end )
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{
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int lpaintedtime = g_paintedtime;
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int endtime = end;
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// resumes playback...
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if ( m_sndBuffers )
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{
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S_TransferStereo16( m_sndBuffers, PAINTBUFFER, lpaintedtime, endtime );
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}
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}
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void CAudioDeviceOpenAL::SpatializeChannel( int volume[CCHANVOLUMES/2], int master_vol, const Vector& sourceDir, float gain, float mono )
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{
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VPROF("CAudioDeviceOpenAL::SpatializeChannel");
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||
S_SpatializeChannel( volume, master_vol, &sourceDir, gain, mono );
|
||
}
|
||
|
||
void CAudioDeviceOpenAL::StopAllSounds( void )
|
||
{
|
||
m_bSoundsShutdown = true;
|
||
alSourceStop( m_Source[0] );
|
||
}
|
||
|
||
|
||
|
||
void CAudioDeviceOpenAL::ApplyDSPEffects( int idsp, portable_samplepair_t *pbuffront, portable_samplepair_t *pbufrear, portable_samplepair_t *pbufcenter, int samplecount )
|
||
{
|
||
//SX_RoomFX( endtime, filter, timefx );
|
||
DSP_Process( idsp, pbuffront, pbufrear, pbufcenter, samplecount );
|
||
}
|
||
|
||
|
||
static uint32 GetOSXSpeakerConfig()
|
||
{
|
||
return 2;
|
||
}
|
||
|
||
static uint32 GetSpeakerConfigForSurroundMode( int surroundMode, const char **pConfigDesc )
|
||
{
|
||
uint32 newSpeakerConfig = 2;
|
||
*pConfigDesc = "stereo speaker";
|
||
return newSpeakerConfig;
|
||
}
|
||
|
||
|
||
|
||
void OnSndSurroundCvarChanged( IConVar *pVar, const char *pOldString, float flOldValue )
|
||
{
|
||
// if the old value is -1, we're setting this from the detect routine for the first time
|
||
// no need to reset the device
|
||
if ( flOldValue == -1 )
|
||
return;
|
||
|
||
// get the user's previous speaker config
|
||
uint32 speaker_config = GetOSXSpeakerConfig();
|
||
|
||
// get the new config
|
||
uint32 newSpeakerConfig = 0;
|
||
const char *speakerConfigDesc = "";
|
||
|
||
ConVarRef var( pVar );
|
||
newSpeakerConfig = GetSpeakerConfigForSurroundMode( var.GetInt(), &speakerConfigDesc );
|
||
// make sure the config has changed
|
||
if (newSpeakerConfig == speaker_config)
|
||
return;
|
||
|
||
// set new configuration
|
||
//SetWindowsSpeakerConfig(newSpeakerConfig);
|
||
|
||
Msg("Speaker configuration has been changed to %s.\n", speakerConfigDesc);
|
||
|
||
// restart sound system so it takes effect
|
||
//g_pSoundServices->RestartSoundSystem();
|
||
}
|
||
|
||
void OnSndSurroundLegacyChanged( IConVar *pVar, const char *pOldString, float flOldValue )
|
||
{
|
||
}
|
||
|
||
#endif // !DEDICATED
|
||
|