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[FreeAudio.mod] Backport NG adjustments #8

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20 changes: 19 additions & 1 deletion .gitignore
100644 → 100755
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Original file line number Diff line number Diff line change
Expand Up @@ -2,5 +2,23 @@
*.x86.a
*.x86.i
*.bak

*.x64.a
*.x64.i
*.arm.a
*.arm.i
*.armeabi.a
*.armeabi.i
*.armeabiv7a.a
*.armeabiv7a.i
*.arm64v8a.a
*.arm64v8a.i
*.js.a
*.js.i
*.armv7.a
*.armv7.i
*.arm64.a
*.arm64.i
.bmx/
*.i2
*.exe
commands.html
33 changes: 17 additions & 16 deletions freeaudio.mod/freeaudio.bmx
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Expand Up @@ -55,15 +55,15 @@ Import "freeaudioglue.cpp"
Import "dsounddevice.cpp"
Import "mmdevice.cpp"
Extern "C"
Function OpenMultiMediaDevice()
Function OpenDirectSoundDevice()
Function OpenMultiMediaDevice:Byte Ptr()
Function OpenDirectSoundDevice:Byte Ptr()
End Extern
?MacOS
Import "-framework AudioUnit"
Import "-framework AudioToolbox"
Import "coreaudiodevice.cpp"
Extern
Function OpenCoreAudioDevice()
Function OpenCoreAudioDevice:Byte Ptr()
End Extern
?Linux
Import "-ldl"
Expand All @@ -74,8 +74,8 @@ Import "pulseaudiodevice.cpp"
'Import "ossdevice.cpp"
Extern "C"
'Function OpenOSSDevice()
Function OpenALSADevice()
Function OpenPulseAudioDevice()
Function OpenALSADevice:Byte Ptr()
Function OpenPulseAudioDevice:Byte Ptr()
End Extern
?

Expand All @@ -88,14 +88,14 @@ Const FA_CHANNELSTATUS_LOOPING=4
Const FA_CHANNELSTATUS_STREAMING=8
Const FA_CHANNELSTATUS_PAUSED=16

Function fa_Reset( audiodevice )
Function fa_Reset( audiodevice:Byte Ptr )
Function fa_Close()
Function fa_CreateSound( length,bits,channels,hertz,samples:Byte Ptr=Null,looping=False )
Function fa_WriteSound( sound,samples:Byte Ptr,length ) 'length really neceesary?
Function fa_FreeSound( sound )
Function fa_CreateSound:Byte Ptr( length,bits,channels,hertz,samples:Byte Ptr=Null,looping=False )
Function fa_WriteSound( sound:Byte Ptr,samples:Byte Ptr,length ) 'length really neceesary?
Function fa_FreeSound( sound:Byte Ptr )
Function fa_AllocChannel()
Function fa_FreeChannel( channel )
Function fa_PlaySound( sound,paused_flag,channel )
Function fa_PlaySound( sound:Byte Ptr,paused_flag,channel )

Function fa_StopChannel( channel )
Function fa_ChannelStatus( channel )
Expand All @@ -110,13 +110,14 @@ Function fa_SetChannelDepth( channel,depth# )
End Extern

Function fa_Init( deviceid )
Local device
Local device:Byte Ptr
?Win32
If deviceid
device=OpenDirectSoundDevice()
Else
device=OpenMultiMediaDevice()
EndIf
Select deviceid
Case 0
device=OpenMultiMediaDevice()
Case 1
device=OpenDirectSoundDevice()
EndSelect
?Linux
Select deviceid
Case 0
Expand Down
10 changes: 5 additions & 5 deletions freeaudio.mod/freeaudioglue.cpp
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Expand Up @@ -22,7 +22,7 @@ int BBCALL fa_Close(){
return 0;
}

int BBCALL fa_PlaySound( int sound,int paused,int ch ){
int BBCALL fa_PlaySound( sample* sound,int paused,int ch ){
output *out;
bool recycle;
if (audio && sound)
Expand Down Expand Up @@ -121,7 +121,7 @@ int BBCALL fa_SetChannelDepth( int channel,float depth ){
return 0;
}

int BBCALL fa_CreateSound( int length,int bits,int channels,int freq,const char *samples,int loop ){
sample* BBCALL fa_CreateSound( int length,int bits,int channels,int freq,const char *samples,int loop ){
sample *sam;

if (io==0) return 0;
Expand All @@ -133,18 +133,18 @@ int BBCALL fa_CreateSound( int length,int bits,int channels,int freq,const char
if( samples ) sam->write((void*)samples,length,0);
}
sam->setloop(loop);
return (int)sam;
return sam;
}

int BBCALL fa_WriteSound( int sound, void *data, int samples){
int BBCALL fa_WriteSound( sample* sound, void *data, int samples){
sample *sam;

if (io==0) return 0;
sam=(sample*)sound;
return sam->write(data,samples,0);
}

int BBCALL fa_FreeSound( int sound ){
int BBCALL fa_FreeSound( sample* sound ){
sample *sam;
if (io==0) return 0;
sam=(sample*)sound;
Expand Down
8 changes: 4 additions & 4 deletions freeaudio.mod/freeaudioglue.h
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Expand Up @@ -16,10 +16,10 @@ extern "C" {
int fa_Reset(struct audiodevice *device);
int fa_Close();

int fa_CreateSound(int samples,int bits,int channels,int freq,const char *buf,int loop);
int fa_WriteSound( int sound, void *data, int samples);
int fa_FreeSound( int sound );
int fa_PlaySound( int sound,int paused,int channel );
sample* fa_CreateSound(int samples,int bits,int channels,int freq,const char *buf,int loop);
int fa_WriteSound( sample* sound, void *data, int samples);
int fa_FreeSound( sample* sound );
int fa_PlaySound( sample* sound,int paused,int channel );

int fa_AllocChannel();
void fa_FreeChannel( int channel );
Expand Down
14 changes: 7 additions & 7 deletions freeaudio.mod/mmdevice.cpp
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Expand Up @@ -14,7 +14,7 @@

extern "C" audiodevice *OpenMultiMediaDevice();

static void __stdcall audioTimerProc( UINT id,UINT msg,DWORD user,DWORD u1,DWORD u2 );
static void __stdcall audioTimerProc( UINT id,UINT msg,DWORD_PTR user,DWORD_PTR u1,DWORD_PTR u2 );

struct pcmsetting
{
Expand Down Expand Up @@ -43,7 +43,7 @@ struct abuffer {
id=n;
hdr->lpData=(LPSTR)data;
hdr->dwBufferLength=size;
hdr->dwUser=(int)this;
hdr->dwUser=(DWORD_PTR)this;
hdr->dwFlags=WHDR_BEGINLOOP|WHDR_ENDLOOP;
hdr->dwLoops=0x7fffffff;
waveOutPrepareHeader(device,hdr,sizeof(WAVEHDR));
Expand Down Expand Up @@ -94,7 +94,7 @@ struct winmmdevice:audiodevice {

int reset()
{
int sz=(iswin98())?2048:1024; //10ms fragment size 20ms for windows98
int sz=2048; // 20ms fragment size
mix=new mixer(sz);
device=0;
buffersize=0;
Expand Down Expand Up @@ -126,13 +126,13 @@ struct winmmdevice:audiodevice {
{
pcmsetting pcm(freq,channels,bits);
if (bits==8) {mode=0;samplesize=1;} else {mode=1;samplesize=2;}
if (waveOutOpen(&device,WAVE_MAPPER,&pcm.wf,0,(long)this,0)) return 1;
if (waveOutOpen(&device,WAVE_MAPPER,&pcm.wf,0,(DWORD_PTR)this,0)) return 1;
buffersize=size;
bnum=0;
mix->freq=freq;
mix->channels=channels;
buffer.init(device,size*samplesize*32,0);
timeSetEvent(5,5,audioTimerProc,(DWORD)this,TIME_ONESHOT);//PERIODIC );
timeSetEvent(5,5,audioTimerProc,(DWORD_PTR)this,TIME_ONESHOT);//PERIODIC );
timeout=0;
playing=1;
lagbuffers=6;
Expand Down Expand Up @@ -195,9 +195,9 @@ struct winmmdevice:audiodevice {
}
};

static void __stdcall audioTimerProc( UINT id,UINT msg,DWORD user,DWORD u1,DWORD u2 ) {
static void __stdcall audioTimerProc( UINT id,UINT msg,DWORD_PTR user,DWORD_PTR u1,DWORD_PTR u2 ) {
((winmmdevice *)user)->flip();
timeSetEvent(5,5,audioTimerProc,(DWORD)user,TIME_ONESHOT);
timeSetEvent(5,5,audioTimerProc,(DWORD_PTR)user,TIME_ONESHOT);
}

audiodevice *OpenMultiMediaDevice() {
Expand Down
36 changes: 31 additions & 5 deletions freeaudio.mod/pulseaudiodevice.cpp
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Expand Up @@ -8,6 +8,8 @@
#include <unistd.h>
#include <pulse/simple.h>
#include <pthread.h>
//to allow access to the AppTitle
#include <brl.mod/blitz.mod/blitz.h>

extern "C" audiodevice *OpenPulseAudioDevice();

Expand All @@ -26,13 +28,34 @@ int LoadPulse(){
return _pulse!=0;
}

void InitPulse(){
int InitPulse(){

if( !_pulse ) return;
if( !_pulse ) return 0;

pulse_pa_simple_new = (pa_simple* (*)(const char *, const char *, pa_stream_direction_t , const char *, const char *, const pa_sample_spec *, const pa_channel_map *, const pa_buffer_attr *, int *))dlsym(_pulse, "pa_simple_new");
pulse_pa_simple_free = (void (*)(pa_simple *))dlsym(_pulse, "pa_simple_free");
pulse_pa_simple_write = (int (*)(pa_simple *, const void *, size_t , int *))dlsym(_pulse, "pa_simple_write");

// try an open a connection to pulse - if it fails, we can presume pulse is not working?

pa_sample_spec spec;
spec.format=PA_SAMPLE_S16LE;
spec.rate=44100;
spec.channels=2;
int error = 0;

char *p=bbStringToUTF8String( bbAppTitle );
pa_simple * simple = pulse_pa_simple_new(NULL,p,PA_STREAM_PLAYBACK,NULL,"playback",&spec,NULL,NULL,&error);
bbMemFree(p);

if (simple) {
pulse_pa_simple_free(simple);

if (!error) {
return 1;
}
}
return 0;
}

}
Expand Down Expand Up @@ -72,7 +95,9 @@ struct pulseaudiodevice:audiodevice{
stereo16.rate=44100;
stereo16.channels=2;

simple=pulse_pa_simple_new(NULL,"freeaudio",PA_STREAM_PLAYBACK,NULL,"playback",&stereo16,NULL,NULL,&error);
char *p=bbStringToUTF8String( bbAppTitle );
simple=pulse_pa_simple_new(NULL,p,PA_STREAM_PLAYBACK,NULL,"playback",&stereo16,NULL,NULL,&error);
bbMemFree(p);
if (simple==NULL) return -1;

mix=new mixer(LINUXFRAG);
Expand Down Expand Up @@ -120,8 +145,9 @@ struct pulseaudiodevice:audiodevice{

audiodevice *OpenPulseAudioDevice(){
if (LoadPulse()) {
InitPulse();
return new pulseaudiodevice();
if (InitPulse()) {
return new pulseaudiodevice();
}
}
return 0;
}
Expand Down