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dosbox-staging/src/hardware/mixer.cpp
krcroft 7375db2a83 Use type-specific host conversion funcs
2020-04-15 14:26:29 +02:00

807 lines
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/*
* Copyright (C) 2002-2020 The DOSBox Team
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with this program; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA.
*/
/*
Remove the sdl code from here and have it handeld in the sdlmain.
That should call the mixer start from there or something.
*/
// #define DEBUG 1
#include <string.h>
#include <sys/types.h>
#include <math.h>
#include <algorithm>
#if defined (WIN32)
//Midi listing
#ifndef WIN32_LEAN_AND_MEAN
#define WIN32_LEAN_AND_MEAN
#endif
#include <windows.h>
#include <mmsystem.h>
#endif
#include <SDL.h>
#include "mem.h"
#include "pic.h"
#include "dosbox.h"
#include "mixer.h"
#include "timer.h"
#include "setup.h"
#include "cross.h"
#include "support.h"
#include "mapper.h"
#include "hardware.h"
#include "programs.h"
#include "midi.h"
#define MIXER_SSIZE 4
//#define MIXER_SHIFT 14
//#define MIXER_REMAIN ((1<<MIXER_SHIFT)-1)
#define MIXER_VOLSHIFT 13
#define FREQ_SHIFT 14
#define FREQ_NEXT ( 1 << FREQ_SHIFT)
#define FREQ_MASK ( FREQ_NEXT -1 )
#define TICK_SHIFT 14
#define TICK_NEXT ( 1 << TICK_SHIFT)
#define TICK_MASK (TICK_NEXT -1)
static INLINE Bit16s MIXER_CLIP(Bits SAMP) {
if (SAMP < MAX_AUDIO) {
if (SAMP > MIN_AUDIO)
return SAMP;
else return MIN_AUDIO;
} else return MAX_AUDIO;
}
static struct {
int32_t work[MIXER_BUFSIZE][2];
//Write/Read pointers for the buffer
Bitu pos,done;
Bitu needed, min_needed, max_needed;
//For every millisecond tick how many samples need to be generated
Bit32u tick_add;
Bit32u tick_counter;
float mastervol[2];
MixerChannel * channels;
bool nosound;
Bit32u freq;
Bit32u blocksize;
//Note: As stated earlier, all sdl code shall rather be in sdlmain
SDL_AudioDeviceID sdldevice;
} mixer;
Bit8u MixTemp[MIXER_BUFSIZE];
MixerChannel::MixerChannel(MIXER_Handler _handler, Bitu _freq, const char * _name) :
// Public member initialization
volmain {0, 0},
next (nullptr),
name (_name),
done (0),
enabled (false),
// Private member initialization
handler (_handler),
freq_add (0),
freq_counter (0),
needed (0),
prev_sample {0, 0},
next_sample {0, 0},
volmul {0, 0},
scale {0, 0},
channel_map {0, 0},
interpolate (false) {
}
MixerChannel * MIXER_AddChannel(MIXER_Handler handler, Bitu freq, const char * name) {
MixerChannel * chan=new MixerChannel(handler, freq, name);
chan->next=mixer.channels;
chan->SetFreq(freq); // also enables 'interpolate' if needed
chan->SetScale(1.0);
chan->SetVolume(1, 1);
chan->MapChannels(0, 1);
chan->Enable(false);
mixer.channels=chan;
return chan;
}
MixerChannel * MIXER_FindChannel(const char * name) {
MixerChannel * chan=mixer.channels;
while (chan) {
if (!strcasecmp(chan->name,name)) break;
chan=chan->next;
}
return chan;
}
void MIXER_DelChannel(MixerChannel* delchan) {
MixerChannel * chan=mixer.channels;
MixerChannel * * where=&mixer.channels;
while (chan) {
if (chan==delchan) {
*where=chan->next;
delete delchan;
return;
}
where=&chan->next;
chan=chan->next;
}
}
static void MIXER_LockAudioDevice(void) {
SDL_LockAudioDevice(mixer.sdldevice);
}
static void MIXER_UnlockAudioDevice(void) {
SDL_UnlockAudioDevice(mixer.sdldevice);
}
void MixerChannel::UpdateVolume(void) {
volmul[0]=(Bits)((1 << MIXER_VOLSHIFT)*scale[0]*volmain[0]*mixer.mastervol[0]);
volmul[1]=(Bits)((1 << MIXER_VOLSHIFT)*scale[1]*volmain[1]*mixer.mastervol[1]);
}
void MixerChannel::SetVolume(float _left,float _right) {
// Allow unconstrained user-defined values
volmain[0] = _left;
volmain[1] = _right;
UpdateVolume();
}
void MixerChannel::SetScale(float f) {
SetScale(f, f);
}
void MixerChannel::SetScale(float _left, float _right) {
// Constrain application-defined volume between 0% and 100%
const float min_volume(0.0);
const float max_volume(1.0);
_left = clamp(_left, min_volume, max_volume);
_right = clamp(_right, min_volume, max_volume);
if (scale[0] != _left || scale[1] != _right) {
scale[0] = _left;
scale[1] = _right;
UpdateVolume();
#ifdef DEBUG
LOG_MSG("MIXER %-7s channel: application changed left and right volumes to %3.0f%% and %3.0f%%, respectively",
name, scale[0] * 100, scale[1] * 100);
#endif
}
}
void MixerChannel::MapChannels(Bit8u _left, Bit8u _right) {
// Constrain mapping to the 0 (left) and 1 (right) channel indexes
const Bit8u min_channel(0);
const Bit8u max_channel(1);
_left = clamp(_left, min_channel, max_channel);
_right = clamp(_right, min_channel, max_channel);
if (channel_map[0] != _left || channel_map[1] != _right) {
channel_map[0] = _left;
channel_map[1] = _right;
#ifdef DEBUG
LOG_MSG("MIXER %-7s channel: application changed audio-channel mapping to left=>%s and right=>%s",
name,
channel_map[0] == 0 ? "left" : "right",
channel_map[1] == 0 ? "left" : "right");
#endif
}
}
void MixerChannel::Enable(bool _yesno) {
if (_yesno==enabled) return;
enabled=_yesno;
if (enabled) {
freq_counter = 0;
MIXER_LockAudioDevice();
if (done<mixer.done) done=mixer.done;
MIXER_UnlockAudioDevice();
}
}
void MixerChannel::SetFreq(Bitu freq) {
freq_add=(freq<<FREQ_SHIFT)/mixer.freq;
if (freq != mixer.freq) {
interpolate = true;
}
else {
interpolate = false;
}
}
void MixerChannel::Mix(Bitu _needed) {
needed=_needed;
while (enabled && needed>done) {
Bitu left = (needed - done);
left *= freq_add;
left = (left >> FREQ_SHIFT) + ((left & FREQ_MASK)!=0);
handler(left);
}
}
void MixerChannel::AddSilence(void) {
if (done<needed) {
done=needed;
//Make sure the next samples are zero when they get switched to prev
next_sample[0] = 0;
next_sample[1] = 0;
//This should trigger an instant request for new samples
freq_counter = FREQ_NEXT;
}
}
template<class Type,bool stereo,bool signeddata,bool nativeorder>
inline void MixerChannel::AddSamples(Bitu len, const Type* data) {
//Position where to write the data
Bitu mixpos = mixer.pos + done;
//Position in the incoming data
Bitu pos = 0;
//Mix and data for the full length
while (1) {
//Does new data need to get read?
while (freq_counter >= FREQ_NEXT) {
//Would this overflow the source data, then it's time to leave
if (pos >= len)
return;
freq_counter -= FREQ_NEXT;
prev_sample[0] = next_sample[0];
if (stereo) {
prev_sample[1] = next_sample[1];
}
if ( sizeof( Type) == 1) {
if (!signeddata) {
if (stereo) {
next_sample[0]=(((Bit8s)(data[pos*2+0] ^ 0x80)) << 8);
next_sample[1]=(((Bit8s)(data[pos*2+1] ^ 0x80)) << 8);
} else {
next_sample[0]=(((Bit8s)(data[pos] ^ 0x80)) << 8);
}
} else {
if (stereo) {
next_sample[0]=(data[pos*2+0] << 8);
next_sample[1]=(data[pos*2+1] << 8);
} else {
next_sample[0]=(data[pos] << 8);
}
}
//16bit and 32bit both contain 16bit data internally
} else {
if (signeddata) {
if (stereo) {
if (nativeorder) {
next_sample[0]=data[pos*2+0];
next_sample[1]=data[pos*2+1];
} else {
if ( sizeof( Type) == 2) {
next_sample[0]=(Bit16s)host_readw((HostPt)&data[pos*2+0]);
next_sample[1]=(Bit16s)host_readw((HostPt)&data[pos*2+1]);
} else {
next_sample[0]=(Bit32s)host_readd((HostPt)&data[pos*2+0]);
next_sample[1]=(Bit32s)host_readd((HostPt)&data[pos*2+1]);
}
}
} else {
if (nativeorder) {
next_sample[0] = data[pos];
} else {
if ( sizeof( Type) == 2) {
next_sample[0]=(Bit16s)host_readw((HostPt)&data[pos]);
} else {
next_sample[0]=(Bit32s)host_readd((HostPt)&data[pos]);
}
}
}
} else {
if (stereo) {
if (nativeorder) {
next_sample[0]=(Bits)data[pos*2+0]-32768;
next_sample[1]=(Bits)data[pos*2+1]-32768;
} else {
if ( sizeof( Type) == 2) {
next_sample[0]=(Bits)host_readw((HostPt)&data[pos*2+0])-32768;
next_sample[1]=(Bits)host_readw((HostPt)&data[pos*2+1])-32768;
} else {
next_sample[0]=(Bits)host_readd((HostPt)&data[pos*2+0])-32768;
next_sample[1]=(Bits)host_readd((HostPt)&data[pos*2+1])-32768;
}
}
} else {
if (nativeorder) {
next_sample[0]=(Bits)data[pos]-32768;
} else {
if ( sizeof( Type) == 2) {
next_sample[0]=(Bits)host_readw((HostPt)&data[pos])-32768;
} else {
next_sample[0]=(Bits)host_readd((HostPt)&data[pos])-32768;
}
}
}
}
}
//This sample has been handled now, increase position
pos++;
}
//Apply the left and right channel mappers only on write[..]
//assignments. This ensures the channels are mapped only once
//(avoiding double-swapping) and also minimizes the places where
//we use our mapping variables as array indexes.
//Note that volumes are independent of the channels mapping.
const Bit8u left_map(channel_map[0]);
const Bit8u right_map(channel_map[1]);
//Where to write
mixpos &= MIXER_BUFMASK;
Bit32s* write = mixer.work[mixpos];
if (!interpolate) {
write[0] += prev_sample[left_map] * volmul[0];
write[1] += (stereo ? prev_sample[right_map] : prev_sample[left_map]) * volmul[1];
}
else {
Bits diff_mul = freq_counter & FREQ_MASK;
Bits sample = prev_sample[left_map] + (((next_sample[left_map] - prev_sample[left_map]) * diff_mul) >> FREQ_SHIFT);
write[0] += sample*volmul[0];
if (stereo) {
sample = prev_sample[right_map] + (((next_sample[right_map] - prev_sample[right_map]) * diff_mul) >> FREQ_SHIFT);
}
write[1] += sample*volmul[1];
}
//Prepare for next sample
freq_counter += freq_add;
mixpos++;
done++;
}
}
void MixerChannel::AddStretched(Bitu len,Bit16s * data) {
if (done >= needed) {
LOG_MSG("Can't add, buffer full");
return;
}
//Target samples this inputs gets stretched into
Bitu outlen = needed - done;
Bitu index = 0;
Bitu index_add = (len << FREQ_SHIFT)/outlen;
Bitu mixpos = mixer.pos + done;
done = needed;
Bitu pos = 0;
while (outlen--) {
Bitu new_pos = index >> FREQ_SHIFT;
if (pos != new_pos) {
pos = new_pos;
//Forward the previous sample
prev_sample[0] = data[0];
data++;
}
Bits diff = data[0] - prev_sample[0];
Bits diff_mul = index & FREQ_MASK;
index += index_add;
mixpos &= MIXER_BUFMASK;
Bits sample = prev_sample[0] + ((diff * diff_mul) >> FREQ_SHIFT);
mixer.work[mixpos][0] += sample * volmul[0];
mixer.work[mixpos][1] += sample * volmul[1];
mixpos++;
}
}
void MixerChannel::AddSamples_m8(Bitu len, const Bit8u * data) {
AddSamples<Bit8u,false,false,true>(len,data);
}
void MixerChannel::AddSamples_s8(Bitu len,const Bit8u * data) {
AddSamples<Bit8u,true,false,true>(len,data);
}
void MixerChannel::AddSamples_m8s(Bitu len,const Bit8s * data) {
AddSamples<Bit8s,false,true,true>(len,data);
}
void MixerChannel::AddSamples_s8s(Bitu len,const Bit8s * data) {
AddSamples<Bit8s,true,true,true>(len,data);
}
void MixerChannel::AddSamples_m16(Bitu len,const Bit16s * data) {
AddSamples<Bit16s,false,true,true>(len,data);
}
void MixerChannel::AddSamples_s16(Bitu len,const Bit16s * data) {
AddSamples<Bit16s,true,true,true>(len,data);
}
void MixerChannel::AddSamples_m16u(Bitu len,const Bit16u * data) {
AddSamples<Bit16u,false,false,true>(len,data);
}
void MixerChannel::AddSamples_s16u(Bitu len,const Bit16u * data) {
AddSamples<Bit16u,true,false,true>(len,data);
}
void MixerChannel::AddSamples_m32(Bitu len,const Bit32s * data) {
AddSamples<Bit32s,false,true,true>(len,data);
}
void MixerChannel::AddSamples_s32(Bitu len,const Bit32s * data) {
AddSamples<Bit32s,true,true,true>(len,data);
}
void MixerChannel::AddSamples_m16_nonnative(Bitu len,const Bit16s * data) {
AddSamples<Bit16s,false,true,false>(len,data);
}
void MixerChannel::AddSamples_s16_nonnative(Bitu len,const Bit16s * data) {
AddSamples<Bit16s,true,true,false>(len,data);
}
void MixerChannel::AddSamples_m16u_nonnative(Bitu len,const Bit16u * data) {
AddSamples<Bit16u,false,false,false>(len,data);
}
void MixerChannel::AddSamples_s16u_nonnative(Bitu len,const Bit16u * data) {
AddSamples<Bit16u,true,false,false>(len,data);
}
void MixerChannel::AddSamples_m32_nonnative(Bitu len,const Bit32s * data) {
AddSamples<Bit32s,false,true,false>(len,data);
}
void MixerChannel::AddSamples_s32_nonnative(Bitu len,const Bit32s * data) {
AddSamples<Bit32s,true,true,false>(len,data);
}
void MixerChannel::FillUp(void) {
MIXER_LockAudioDevice();
if (!enabled || done<mixer.done) {
MIXER_UnlockAudioDevice();
return;
}
float index=PIC_TickIndex();
Mix((Bitu)(index*mixer.needed));
MIXER_UnlockAudioDevice();
}
extern bool ticksLocked;
static inline bool Mixer_irq_important(void) {
/* In some states correct timing of the irqs is more important then
* non stuttering audo */
return (ticksLocked || (CaptureState & (CAPTURE_WAVE|CAPTURE_VIDEO)));
}
static Bit32u calc_tickadd(Bit32u freq) {
#if TICK_SHIFT >16
Bit64u freq64 = static_cast<Bit64u>(freq);
freq64 = (freq64<<TICK_SHIFT)/1000;
Bit32u r = static_cast<Bit32u>(freq64);
return r;
#else
return (freq<<TICK_SHIFT)/1000;
#endif
}
/* Mix a certain amount of new samples */
static void MIXER_MixData(Bitu needed) {
MixerChannel * chan=mixer.channels;
while (chan) {
chan->Mix(needed);
chan=chan->next;
}
if (CaptureState & (CAPTURE_WAVE|CAPTURE_VIDEO)) {
int16_t convert[1024][2];
const size_t added = std::min<size_t>(needed - mixer.done, 1024);
size_t readpos = (mixer.pos + mixer.done) & MIXER_BUFMASK;
for (size_t i = 0; i < added; i++) {
const int32_t sample_1 = mixer.work[readpos][0] >> MIXER_VOLSHIFT;
const int32_t sample_2 = mixer.work[readpos][1] >> MIXER_VOLSHIFT;
const int16_t s1 = MIXER_CLIP(sample_1);
const int16_t s2 = MIXER_CLIP(sample_2);
convert[i][0] = host_to_le16(s1);
convert[i][1] = host_to_le16(s2);
readpos = (readpos + 1) & MIXER_BUFMASK;
}
CAPTURE_AddWave(mixer.freq, added, reinterpret_cast<int16_t*>(convert));
}
//Reset the the tick_add for constant speed
if( Mixer_irq_important() )
mixer.tick_add = calc_tickadd(mixer.freq);
mixer.done = needed;
}
static void MIXER_Mix(void) {
MIXER_LockAudioDevice();
MIXER_MixData(mixer.needed);
mixer.tick_counter += mixer.tick_add;
mixer.needed+=(mixer.tick_counter >> TICK_SHIFT);
mixer.tick_counter &= TICK_MASK;
MIXER_UnlockAudioDevice();
}
static void MIXER_Mix_NoSound(void) {
MIXER_MixData(mixer.needed);
/* Clear piece we've just generated */
for (Bitu i=0;i<mixer.needed;i++) {
mixer.work[mixer.pos][0]=0;
mixer.work[mixer.pos][1]=0;
mixer.pos=(mixer.pos+1)&MIXER_BUFMASK;
}
/* Reduce count in channels */
for (MixerChannel * chan=mixer.channels;chan;chan=chan->next) {
if (chan->done>mixer.needed) chan->done-=mixer.needed;
else chan->done=0;
}
/* Set values for next tick */
mixer.tick_counter += mixer.tick_add;
mixer.needed = (mixer.tick_counter >> TICK_SHIFT);
mixer.tick_counter &= TICK_MASK;
mixer.done=0;
}
static void SDLCALL MIXER_CallBack(void * userdata, Uint8 *stream, int len) {
memset(stream, 0, len);
Bitu need=(Bitu)len/MIXER_SSIZE;
Bit16s * output=(Bit16s *)stream;
Bitu reduce;
Bitu pos;
//Local resampling counter to manipulate the data when sending it off to the callback
Bitu index, index_add;
Bits sample;
/* Enough room in the buffer ? */
if (mixer.done < need) {
// LOG_MSG("Full underrun need %d, have %d, min %d", need, mixer.done, mixer.min_needed);
if((need - mixer.done) > (need >>7) ) //Max 1 procent stretch.
return;
reduce = mixer.done;
index_add = (reduce << TICK_SHIFT) / need;
mixer.tick_add = calc_tickadd(mixer.freq+mixer.min_needed);
} else if (mixer.done < mixer.max_needed) {
Bitu left = mixer.done - need;
if (left < mixer.min_needed) {
if( !Mixer_irq_important() ) {
Bitu needed = mixer.needed - need;
Bitu diff = (mixer.min_needed>needed?mixer.min_needed:needed) - left;
mixer.tick_add = calc_tickadd(mixer.freq+(diff*3));
left = 0; //No stretching as we compensate with the tick_add value
} else {
left = (mixer.min_needed - left);
left = 1 + (2*left) / mixer.min_needed; //left=1,2,3
}
// LOG_MSG("needed underrun need %d, have %d, min %d, left %d", need, mixer.done, mixer.min_needed, left);
reduce = need - left;
index_add = (reduce << TICK_SHIFT) / need;
} else {
reduce = need;
index_add = (1 << TICK_SHIFT);
// LOG_MSG("regular run need %d, have %d, min %d, left %d", need, mixer.done, mixer.min_needed, left);
/* Mixer tick value being updated:
* 3 cases:
* 1) A lot too high. >division by 5. but maxed by 2* min to prevent too fast drops.
* 2) A little too high > division by 8
* 3) A little to nothing above the min_needed buffer > go to default value
*/
Bitu diff = left - mixer.min_needed;
if(diff > (mixer.min_needed<<1)) diff = mixer.min_needed<<1;
if(diff > (mixer.min_needed>>1))
mixer.tick_add = calc_tickadd(mixer.freq-(diff/5));
else if (diff > (mixer.min_needed>>2))
mixer.tick_add = calc_tickadd(mixer.freq-(diff>>3));
else
mixer.tick_add = calc_tickadd(mixer.freq);
}
} else {
/* There is way too much data in the buffer */
// LOG_MSG("overflow run need %d, have %d, min %d", need, mixer.done, mixer.min_needed);
if (mixer.done > MIXER_BUFSIZE)
index_add = MIXER_BUFSIZE - 2*mixer.min_needed;
else
index_add = mixer.done - 2*mixer.min_needed;
index_add = (index_add << TICK_SHIFT) / need;
reduce = mixer.done - 2* mixer.min_needed;
mixer.tick_add = calc_tickadd(mixer.freq-(mixer.min_needed/5));
}
/* Reduce done count in all channels */
for (MixerChannel * chan=mixer.channels;chan;chan=chan->next) {
if (chan->done>reduce) chan->done-=reduce;
else chan->done=0;
}
// Reset mixer.tick_add when irqs are important
if( Mixer_irq_important() )
mixer.tick_add = calc_tickadd(mixer.freq);
mixer.done -= reduce;
mixer.needed -= reduce;
pos = mixer.pos;
mixer.pos = (mixer.pos + reduce) & MIXER_BUFMASK;
index = 0;
if(need != reduce) {
while (need--) {
Bitu i = (pos + (index >> TICK_SHIFT )) & MIXER_BUFMASK;
index += index_add;
sample=mixer.work[i][0]>>MIXER_VOLSHIFT;
*output++=MIXER_CLIP(sample);
sample=mixer.work[i][1]>>MIXER_VOLSHIFT;
*output++=MIXER_CLIP(sample);
}
/* Clean the used buffer */
while (reduce--) {
pos &= MIXER_BUFMASK;
mixer.work[pos][0]=0;
mixer.work[pos][1]=0;
pos++;
}
} else {
while (reduce--) {
pos &= MIXER_BUFMASK;
sample=mixer.work[pos][0]>>MIXER_VOLSHIFT;
*output++=MIXER_CLIP(sample);
sample=mixer.work[pos][1]>>MIXER_VOLSHIFT;
*output++=MIXER_CLIP(sample);
mixer.work[pos][0]=0;
mixer.work[pos][1]=0;
pos++;
}
}
}
static void MIXER_Stop(Section* sec) {
}
class MIXER : public Program {
public:
void MakeVolume(char * scan,float & vol0,float & vol1) {
Bitu w=0;
bool db=(toupper(*scan)=='D');
if (db) scan++;
while (*scan) {
if (*scan==':') {
++scan;w=1;
}
char * before=scan;
float val=(float)strtod(scan,&scan);
if (before==scan) {
++scan;continue;
}
if (!db) val/=100;
else val=powf(10.0f,(float)val/20.0f);
if (val<0) val=1.0f;
if (!w) {
vol0=val;
} else {
vol1=val;
}
}
if (!w) vol1=vol0;
}
void Run(void) {
if(cmd->FindExist("/LISTMIDI")) {
ListMidi();
return;
}
if (cmd->FindString("MASTER",temp_line,false)) {
MakeVolume((char *)temp_line.c_str(),mixer.mastervol[0],mixer.mastervol[1]);
}
MixerChannel * chan = mixer.channels;
while (chan) {
if (cmd->FindString(chan->name,temp_line,false)) {
MakeVolume((char *)temp_line.c_str(),chan->volmain[0],chan->volmain[1]);
}
chan->UpdateVolume();
chan = chan->next;
}
if (cmd->FindExist("/NOSHOW")) return;
WriteOut("Channel Main Main(dB)\n");
ShowVolume("MASTER",mixer.mastervol[0],mixer.mastervol[1]);
for (chan = mixer.channels;chan;chan = chan->next)
ShowVolume(chan->name,chan->volmain[0],chan->volmain[1]);
}
private:
void ShowVolume(const char * name,float vol0,float vol1) {
WriteOut("%-8s %3.0f:%-3.0f %+3.2f:%-+3.2f \n",name,
vol0*100,vol1*100,
20*log(vol0)/log(10.0f),20*log(vol1)/log(10.0f)
);
}
void ListMidi(){
if(midi.handler) midi.handler->ListAll(this);
};
};
static void MIXER_ProgramStart(Program * * make) {
*make=new MIXER;
}
MixerChannel* MixerObject::Install(MIXER_Handler handler,Bitu freq,const char * name){
if(!installed) {
if(strlen(name) > 31) E_Exit("Too long mixer channel name");
safe_strncpy(m_name,name,32);
installed = true;
return MIXER_AddChannel(handler,freq,name);
} else {
E_Exit("already added mixer channel.");
return 0; //Compiler happy
}
}
MixerObject::~MixerObject(){
if(!installed) return;
MIXER_DelChannel(MIXER_FindChannel(m_name));
}
void MIXER_Init(Section* sec) {
sec->AddDestroyFunction(&MIXER_Stop);
Section_prop * section=static_cast<Section_prop *>(sec);
/* Read out config section */
mixer.freq=section->Get_int("rate");
mixer.nosound=section->Get_bool("nosound");
mixer.blocksize=section->Get_int("blocksize");
/* Initialize the internal stuff */
mixer.channels=0;
mixer.pos=0;
mixer.done=0;
memset(mixer.work,0,sizeof(mixer.work));
mixer.mastervol[0]=1.0f;
mixer.mastervol[1]=1.0f;
/* Start the Mixer using SDL Sound at 22 khz */
SDL_AudioSpec spec;
SDL_AudioSpec obtained;
spec.freq=mixer.freq;
spec.format=AUDIO_S16SYS;
spec.channels=2;
spec.callback=MIXER_CallBack;
spec.userdata=NULL;
spec.samples=(Uint16)mixer.blocksize;
mixer.tick_counter=0;
if (mixer.nosound) {
LOG_MSG("MIXER: No Sound Mode Selected.");
mixer.tick_add=calc_tickadd(mixer.freq);
TIMER_AddTickHandler(MIXER_Mix_NoSound);
} else if ((mixer.sdldevice = SDL_OpenAudioDevice(NULL, 0, &spec, &obtained, SDL_AUDIO_ALLOW_FREQUENCY_CHANGE)) ==0 ) {
mixer.nosound = true;
LOG_MSG("MIXER: Can't open audio: %s , running in nosound mode.",SDL_GetError());
mixer.tick_add=calc_tickadd(mixer.freq);
TIMER_AddTickHandler(MIXER_Mix_NoSound);
} else {
if((static_cast<Bit16s>(mixer.freq) != obtained.freq) || (mixer.blocksize != obtained.samples))
LOG_MSG("MIXER: Got different values from SDL: freq %d, blocksize %d",obtained.freq,obtained.samples);
mixer.freq=obtained.freq;
mixer.blocksize=obtained.samples;
mixer.tick_add=calc_tickadd(mixer.freq);
TIMER_AddTickHandler(MIXER_Mix);
SDL_PauseAudioDevice(mixer.sdldevice, 0);
}
mixer.min_needed=section->Get_int("prebuffer");
if (mixer.min_needed>100) mixer.min_needed=100;
mixer.min_needed=(mixer.freq*mixer.min_needed)/1000;
mixer.max_needed=mixer.blocksize * 2 + 2*mixer.min_needed;
mixer.needed=mixer.min_needed+1;
PROGRAMS_MakeFile("MIXER.COM",MIXER_ProgramStart);
}
void MIXER_CloseAudioDevice(void) {
if (!mixer.nosound) {
if (mixer.sdldevice != 0) {
SDL_CloseAudioDevice(mixer.sdldevice);
mixer.sdldevice = 0;
}
}
}
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