mirror of
https://github.com/nillerusr/source-engine.git
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575 lines
17 KiB
C++
575 lines
17 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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#if !DEDICATED
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#include "tier0/dynfunction.h"
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#include "video//ivideoservices.h"
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#include "../../sys_dll.h"
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// prevent some conflicts in SDL headers...
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#undef M_PI
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#include <stdint.h>
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#ifndef _STDINT_H_
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#define _STDINT_H_ 1
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#endif
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#include "SDL.h"
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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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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 nSlot,*/ int volume[6], int master_vol, const Vector *psourceDir, float gain, float mono );
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// 64K is about 1/3 second at 16-bit, stereo, 44100 Hz
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// 44k: UNDONE - need to double buffers now that we're playing back at 44100?
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#define WAV_BUFFERS 64
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#define WAV_MASK (WAV_BUFFERS - 1)
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#define WAV_BUFFER_SIZE 0x0400
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#if 0
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#define debugsdl printf
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#else
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static inline void debugsdl(const char *fmt, ...) {}
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#endif
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//-----------------------------------------------------------------------------
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//
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// NOTE: This only allows 16-bit, stereo wave out (!!! FIXME: but SDL supports 7.1, etc, too!)
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//
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//-----------------------------------------------------------------------------
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class CAudioDeviceSDLAudio : public CAudioDeviceBase
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{
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public:
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CAudioDeviceSDLAudio();
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virtual ~CAudioDeviceSDLAudio();
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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 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 nSlot, 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 "SDL"; }
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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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SDL_AudioDeviceID m_devId;
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static void SDLCALL AudioCallbackEntry(void *userdata, Uint8 * stream, int len);
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void AudioCallback(Uint8 *stream, int len);
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void OpenWaveOut( void );
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void CloseWaveOut( void );
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void AllocateOutputBuffers();
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void FreeOutputBuffers();
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bool ValidWaveOut( void ) const;
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int m_deviceSampleCount;
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int m_buffersSent;
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int m_pauseCount;
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int m_readPos;
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int m_partialWrite;
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// Memory for the wave data
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uint8_t *m_pBuffer;
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};
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static CAudioDeviceSDLAudio *g_wave = NULL;
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//-----------------------------------------------------------------------------
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// Constructor (just lookup SDL entry points, real work happens in this->Init())
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//-----------------------------------------------------------------------------
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CAudioDeviceSDLAudio::CAudioDeviceSDLAudio()
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{
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m_devId = 0;
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}
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//-----------------------------------------------------------------------------
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// Destructor. Make sure our global pointer gets set to NULL.
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//-----------------------------------------------------------------------------
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CAudioDeviceSDLAudio::~CAudioDeviceSDLAudio()
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{
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g_wave = NULL;
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}
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//-----------------------------------------------------------------------------
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// Class factory
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//-----------------------------------------------------------------------------
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IAudioDevice *Audio_CreateSDLAudioDevice( void )
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{
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if ( !g_wave )
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{
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g_wave = new CAudioDeviceSDLAudio;
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Assert( g_wave );
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}
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if ( g_wave && !g_wave->Init() )
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{
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delete g_wave;
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g_wave = NULL;
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}
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return g_wave;
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}
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//-----------------------------------------------------------------------------
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// Init, shutdown
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//-----------------------------------------------------------------------------
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bool CAudioDeviceSDLAudio::Init( void )
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{
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// If we've already got a device open, then return. This allows folks to call
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// Audio_CreateSDLAudioDevice() multiple times. CloseWaveOut() will free the
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// device, and set m_devId to 0.
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if( m_devId )
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return true;
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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_pauseCount = 0;
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m_pBuffer = NULL;
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m_readPos = 0;
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m_partialWrite = 0;
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m_devId = 0;
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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 CAudioDeviceSDLAudio::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 CAudioDeviceSDLAudio::ValidWaveOut( void ) const
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{
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return m_devId != 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 CAudioDeviceSDLAudio::OpenWaveOut( void )
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{
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debugsdl("SDLAUDIO: OpenWaveOut...\n");
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#ifndef WIN32
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char appname[ 256 ];
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KeyValues *modinfo = new KeyValues( "ModInfo" );
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if ( modinfo->LoadFromFile( g_pFileSystem, "gameinfo.txt" ) )
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Q_strncpy( appname, modinfo->GetString( "game" ), sizeof( appname ) );
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else
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Q_strncpy( appname, "Source1 Game", sizeof( appname ) );
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modinfo->deleteThis();
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modinfo = NULL;
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// Set these environment variables, in case we're using PulseAudio.
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setenv("PULSE_PROP_application.name", appname, 1);
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setenv("PULSE_PROP_media.role", "game", 1);
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#endif
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// !!! FIXME: specify channel map, etc
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// !!! FIXME: set properties (role, icon, etc).
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//#define SDLAUDIO_FAIL(fnstr) do { DevWarning(fnstr " failed"); CloseWaveOut(); return; } while (false)
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//#define SDLAUDIO_FAIL(fnstr) do { printf("SDLAUDIO: " fnstr " failed: %s\n", SDL_GetError ? SDL_GetError() : "???"); CloseWaveOut(); return; } while (false)
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#define SDLAUDIO_FAIL(fnstr) do { const char *err = SDL_GetError(); printf("SDLAUDIO: " fnstr " failed: %s\n", err ? err : "???"); CloseWaveOut(); return; } while (false)
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if (!SDL_WasInit(SDL_INIT_AUDIO))
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{
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if (SDL_InitSubSystem(SDL_INIT_AUDIO))
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SDLAUDIO_FAIL("SDL_InitSubSystem(SDL_INIT_AUDIO)");
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}
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debugsdl("SDLAUDIO: Using SDL audio target '%s'\n", SDL_GetCurrentAudioDriver());
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// Open an audio device...
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// !!! FIXME: let user specify a device?
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// !!! FIXME: we can handle quad, 5.1, 7.1, etc here.
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SDL_AudioSpec desired, obtained;
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memset(&desired, '\0', sizeof (desired));
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desired.freq = SOUND_DMA_SPEED;
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desired.format = AUDIO_S16SYS;
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desired.channels = 2;
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desired.samples = 2048;
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desired.callback = &CAudioDeviceSDLAudio::AudioCallbackEntry;
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desired.userdata = this;
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m_devId = SDL_OpenAudioDevice(NULL, 0, &desired, &obtained, SDL_AUDIO_ALLOW_ANY_CHANGE);
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if (!m_devId)
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SDLAUDIO_FAIL("SDL_OpenAudioDevice()");
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#undef SDLAUDIO_FAIL
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// We're now ready to feed audio data to SDL!
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AllocateOutputBuffers();
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SDL_PauseAudioDevice(m_devId, 0);
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#if defined( BINK_VIDEO ) && defined( LINUX )
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// Tells Bink to use SDL for its audio decoding
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if ( g_pVideo != NULL)
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{
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g_pVideo->SoundDeviceCommand( VideoSoundDeviceOperation::SET_SDL_PARAMS, NULL, (void *)&obtained );
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}
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#endif
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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 CAudioDeviceSDLAudio::CloseWaveOut( void )
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{
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// none of these SDL_* functions are available to call if this is false.
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if (m_devId)
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{
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SDL_CloseAudioDevice(m_devId);
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m_devId = 0;
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}
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SDL_QuitSubSystem(SDL_INIT_AUDIO);
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FreeOutputBuffers();
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}
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//-----------------------------------------------------------------------------
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// Allocate output buffers
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//-----------------------------------------------------------------------------
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void CAudioDeviceSDLAudio::AllocateOutputBuffers()
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{
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// Allocate and lock memory for the waveform data.
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const int nBufferSize = WAV_BUFFER_SIZE * WAV_BUFFERS;
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m_pBuffer = new uint8_t[nBufferSize];
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memset(m_pBuffer, '\0', nBufferSize);
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m_readPos = 0;
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m_partialWrite = 0;
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m_deviceSampleCount = nBufferSize / DeviceSampleBytes();
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}
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//-----------------------------------------------------------------------------
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// Free output buffers
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//-----------------------------------------------------------------------------
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void CAudioDeviceSDLAudio::FreeOutputBuffers()
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{
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delete[] m_pBuffer;
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m_pBuffer = NULL;
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}
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//-----------------------------------------------------------------------------
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// Mixing setup
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//-----------------------------------------------------------------------------
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int CAudioDeviceSDLAudio::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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void CAudioDeviceSDLAudio::AudioCallbackEntry(void *userdata, Uint8 *stream, int len)
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{
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((CAudioDeviceSDLAudio *) userdata)->AudioCallback(stream, len);
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}
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void CAudioDeviceSDLAudio::AudioCallback(Uint8 *stream, int len)
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{
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if (!m_devId)
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{
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debugsdl("SDLAUDIO: uhoh, no audio device!\n");
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return; // can this even happen?
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}
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const int totalWriteable = len;
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#if defined( BINK_VIDEO ) && defined( LINUX )
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Uint8 *stream_orig = stream;
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#endif
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debugsdl("SDLAUDIO: writable size is %d.\n", totalWriteable);
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Assert(len <= (WAV_BUFFERS * WAV_BUFFER_SIZE));
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while (len > 0)
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{
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// spaceAvailable == bytes before we overrun the end of the ring buffer.
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const int spaceAvailable = ((WAV_BUFFERS * WAV_BUFFER_SIZE) - m_readPos);
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const int writeLen = (len < spaceAvailable) ? len : spaceAvailable;
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if (writeLen > 0)
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{
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const uint8_t *buf = m_pBuffer + m_readPos;
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debugsdl("SDLAUDIO: Writing %d bytes...\n", writeLen);
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#if 0
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static FILE *io = NULL;
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if (io == NULL) io = fopen("dumpplayback.raw", "wb");
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if (io != NULL) { fwrite(buf, writeLen, 1, io); fflush(io); }
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#endif
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memcpy(stream, buf, writeLen);
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stream += writeLen;
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len -= writeLen;
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Assert(len >= 0);
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}
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m_readPos = len ? 0 : (m_readPos + writeLen); // if still bytes to write to stream, we're rolling around the ring buffer.
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}
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#if defined( BINK_VIDEO ) && defined( LINUX )
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// Mix in Bink movie audio if that stuff is playing.
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if ( g_pVideo != NULL)
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{
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g_pVideo->SoundDeviceCommand( VideoSoundDeviceOperation::SDLMIXER_CALLBACK, (void *)stream_orig, (void *)&totalWriteable );
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}
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#endif
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// Translate between bytes written and buffers written.
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m_partialWrite += totalWriteable;
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m_buffersSent += m_partialWrite / WAV_BUFFER_SIZE;
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m_partialWrite %= WAV_BUFFER_SIZE;
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}
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//-----------------------------------------------------------------------------
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// Actually performs the mixing
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//-----------------------------------------------------------------------------
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void CAudioDeviceSDLAudio::PaintEnd( void )
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{
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debugsdl("SDLAUDIO: PaintEnd...\n");
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#if 0 // !!! FIXME: this is the 1.3 headers, but not implemented yet in SDL.
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if (SDL_AudioDeviceConnected(m_devId) != 1)
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{
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debugsdl("SDLAUDIO: Audio device was disconnected!\n");
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Shutdown();
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}
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#endif
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}
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int CAudioDeviceSDLAudio::GetOutputPosition( void )
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{
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return (m_readPos >> SAMPLE_16BIT_SHIFT)/DeviceChannels();
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}
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//-----------------------------------------------------------------------------
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// Pausing
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//-----------------------------------------------------------------------------
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void CAudioDeviceSDLAudio::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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debugsdl("SDLAUDIO: PAUSE\n");
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SDL_PauseAudioDevice(m_devId, 1);
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}
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}
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void CAudioDeviceSDLAudio::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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if (m_pauseCount == 0)
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{
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debugsdl("SDLAUDIO: UNPAUSE\n");
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SDL_PauseAudioDevice(m_devId, 0);
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}
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}
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}
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bool CAudioDeviceSDLAudio::IsActive( void )
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{
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return ( m_pauseCount == 0 );
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}
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float CAudioDeviceSDLAudio::MixDryVolume( void )
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{
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return 0;
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}
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bool CAudioDeviceSDLAudio::Should3DMix( void )
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{
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return false;
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}
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void CAudioDeviceSDLAudio::ClearBuffer( void )
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{
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int clear;
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if ( !m_pBuffer )
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return;
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clear = 0;
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Q_memset(m_pBuffer, clear, DeviceSampleCount() * DeviceSampleBytes() );
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}
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void CAudioDeviceSDLAudio::MixBegin( int sampleCount )
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{
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MIX_ClearAllPaintBuffers( sampleCount, false );
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}
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void CAudioDeviceSDLAudio::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 CAudioDeviceSDLAudio::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 CAudioDeviceSDLAudio::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 CAudioDeviceSDLAudio::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 CAudioDeviceSDLAudio::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 CAudioDeviceSDLAudio::ChannelReset( int entnum, int channelIndex, float distanceMod )
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{
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}
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void CAudioDeviceSDLAudio::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_pBuffer )
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{
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S_TransferStereo16( m_pBuffer, PAINTBUFFER, lpaintedtime, endtime );
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}
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}
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void CAudioDeviceSDLAudio::SpatializeChannel( int nSlot, int volume[CCHANVOLUMES/2], int master_vol, const Vector& sourceDir, float gain, float mono )
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{
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VPROF("CAudioDeviceSDLAudio::SpatializeChannel");
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S_SpatializeChannel( /*nSlot,*/ volume, master_vol, &sourceDir, gain, mono );
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}
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void CAudioDeviceSDLAudio::StopAllSounds( void )
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{
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}
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void CAudioDeviceSDLAudio::ApplyDSPEffects( int idsp, portable_samplepair_t *pbuffront, portable_samplepair_t *pbufrear, portable_samplepair_t *pbufcenter, int samplecount )
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{
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//SX_RoomFX( endtime, filter, timefx );
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DSP_Process( idsp, pbuffront, pbufrear, pbufcenter, samplecount );
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}
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#endif // !DEDICATED
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