This is Play.m in view mode; [Download] [Up]
// Play, a sound player Object by J. Laroche. June 1992
// Version 2.0
#import "Play.h"
#import<libc.h>
#define Error(A,B) if((A)) { if (delegate && [delegate \
respondsTo:@selector(error:)]) \
{ strcpy(errorMessage,B); \
[delegate error:errorMessage];} \
fprintf(stderr,"%s\n", B); \
SNDRelease(SND_ACCESS_OUT|SND_ACCESS_DSP,dev_port,owner_port); \
return -1;}
//#define Error(A,B) if((A)) { printf("%s \n",B); return self;}
#define DMASIZE 2048
#define MEMMAX 4000
#define WRITE_TAG 0
#define READ_BUF_SIZE (vm_page_size / BYTES_PER_16BIT)
static void HandleDSPMessage(msg_header_t *msg, void *userData)
{
[(Play *)userData playCompleted:0];
}
static void PlayMonitor(DPSTimedEntry tag, double now, char *userData)
{
[(Play *)userData playMonitor];
}
@implementation Play:Object
//------------ Create a new Play object.
+ new
{
self = [super new];
stereo = 0;
verbose = 1;
CHANNEL = MONO;
AES = 0;
INIT = 0;
ster = 0;
U = 0; D = 0; K = 0; Filter_length = 0;
S = 0;
low_water = 48*1024;
high_water = 512*1024;
thresh = 0.003;
offset=0.005;
max_order = (int) (MEMMAX / 150);
status = IDLE;
S_rateIn = 16000;
errorMessage = calloc(1024,sizeof(char));
WAIT = NO;
gain = 0.9;
return self;
}
- setDelegate:anObject
{
delegate = anObject;
return self;
}
- setSamplingRate:(int)aSamplingRate
{
S_rateIn = aSamplingRate;
return self;
}
////////// - setOutput:(int)anOutput. Sets output to desired output:
//// AES_HIGH : ssi port, to A/D64X converters, 48kHz, digital.
//// AES_LOW : ssi port, to A/D64X converters, 44.1kHz, digital.
//// 0 : NeXT Dacs.
- setOutput:(int)anOutput
{
AES = anOutput;
return self;
}
////////////////////////// - setMonoMode:(int)aMode. Sets sound playing mode.
///// MONO or STEREO.
- setMonoMode:(int)aMode
{
CHANNEL = aMode;
stereo = (CHANNEL != MONO);
return self;
}
////////////// - playSound:(SNDSoundStruct*)aSound:(int)begin:(int)length
///// Starts playing length samples of sound, starting from begin.
///// When finished, sends the delegate a didPlay message.
- (int) playSound:(SNDSoundStruct*)aSound:(int)begin:(int)length
{
int i,j;
static int AESStat = 0;
short *beginSound, *endSound;
int beginPage, endPage, pageNumber;
//Sound *DSPprogram;
if(status != IDLE) return -1;
if(aSound == 0) Error(1,"The sound file is null!");
length = MIN(length*(1+stereo),aSound->dataSize/sizeof(short) - begin);
if(aSound->dataSize/sizeof(short) < begin)
Error(1,"Sound not long enough for selected duration");
first_samp = begin;
size_samp = length;
if(!AES) INIT = 0;
else if(AES != AESStat) {INIT = 1; AESStat = AES;}
else INIT = 0;
k_err = SNDAcquire(SND_ACCESS_OUT|SND_ACCESS_DSP,0,0,0,
NULL_NEGOTIATION_FUN,0,&dev_port,&owner_port);
Error(k_err,"DSP or DACs Busy, or access to devices denied\n Are you using another machine's DSP?");
k_err = SNDReset(SND_ACCESS_OUT|SND_ACCESS_DSP,dev_port,owner_port);
Error(k_err,"Error during initialization.");
k_err = snddriver_get_dsp_cmd_port(dev_port,owner_port,&cmd_port);
// Here's an important setting. It appears that this size make the
// whole thing work. others don't. I don't know why, but it has
// something to do with the ramp created by the snd driver.
k_err = snddriver_set_sndout_bufsize(dev_port,owner_port,512);
k_err = snddriver_set_sndout_bufcount(dev_port,owner_port,2);
Error(k_err,"Error during initialization.");
// From now on, we consider we're playing, even if it's not true.
/////////////// Calculate Sampling rate conversion param. //////////////////
S = ((AES == AES_LOW || AES == 0)? 44100 : 48000);
[self calculateRatio:(float)S_rateIn/(float)S:rat:thresh];
D = rat[0];
U = rat[1];
if(D == 0 || U == 0) Error(1,"Weird sampling rate, can't play sound.");
///////////// Examine sound file, set stereo conversion param. ///////////////
if(verbose)
{
printf("Playing\t\t%s\n",file[0]);
if(AES) printf("Output:\t\tA/D64X AES/EBU, %skHz\n",
((AES == AES_LOW) ? "44.1" : "48"));
else printf("Output:\t\t NeXT D/A converters\n");
printf("Sampling rate:\t%d\n", S_rateIn);
switch(CHANNEL)
{
case STEREO : printf("Mode: \t\tStereo\n"); break ;
case MONO : printf("Mode:\t\tMono\n"); break ;
}
}
/////////////// Calculate FIR Filter characteristics //////////////////
if(U == 1 && D == 1) K = 3;
else
// if(AES == 0) K = MIN((int)(40*(2-stereo)*U/D), MIN(MEMMAX/U,60));
// else K = MIN((int)(20*(2-stereo)*U/D), MIN(MEMMAX/U,40));
K = MIN((int)(55*(2-stereo)),MEMMAX/U);
K = MIN(K,250);
if(verbose)
printf("Filter Length: %d, Up-factor %d, Down-factor %d\n",K,U,D);
Filter_length = (((K*U) % 2)? K*U : K*U+1);
para[0] = U;
para[1] = D;
para[2] = K-1;
para[3] = Filter_length;
/////////////// Initialize sound and dsp driver //////////////////
protocol = SNDDRIVER_DSP_PROTO_RAW;
k_err = snddriver_stream_setup(dev_port, owner_port,
SNDDRIVER_DMA_STREAM_TO_DSP,
DMASIZE, 2,
low_water, high_water,
&protocol, &write_port);
Error(k_err,"Error during Stream Set-up");
k_err = snddriver_stream_setup(dev_port, owner_port,
SNDDRIVER_STREAM_DSP_TO_SNDOUT_44,
DMASIZE, 2,
low_water, high_water,
&protocol, &read_port);
Error(k_err,"Error during Stream Set-up");
k_err = snddriver_dsp_protocol(dev_port, owner_port, protocol);
k_err = port_allocate(task_self(),&reply_port);
/////////////// Get DSP program and boot DSP. //////////////////
dspStruct = (SNDSoundStruct *)getsectdata("__SND", "play.snd",&i);
Error(!dspStruct,"Cannot find DSP program!\nSomething's wrong");
// k_err = SNDReadDSPfile("play.lod", &dspStruct,(char*)&i);
// Error(k_err,"Cannot find DSP program! Something's wrong");
k_err = SNDBootDSP(dev_port, owner_port, dspStruct);
Error(k_err,"Cannot boot DSP! Something's wrong");
/////////////// Send stereo param and filter coeff to DSP. //////////////////
ster = stereo + 2 + 4*((AES != 0)) + 8*((AES == AES_HIGH)) +
16*(INIT);
k_err = snddriver_dsp_write(cmd_port,&ster,1,sizeof(int),
SNDDRIVER_LOW_PRIORITY);
filtre = (int*) calloc(Filter_length,sizeof(int));
cut = 3.14159265 * (1/(float) MAX(U,D) - offset);
if(cut <= 0)
{
printf("Problem with the filter's cut-off frequency because U or D \
are too large, \nyou might get aliasing\n");
cut = 3.14159265 * (1/(float) MAX(U,D));
}
if(U == 1 && D == 1)
{
filtre[0] = filtre[2] = 0;
filtre[1] = 8388607 * gain;
}
else [self calculateFilter:filtre:Filter_length:cut: U];
k_err = snddriver_dsp_write(cmd_port,para,4,sizeof(int),
SNDDRIVER_LOW_PRIORITY);
k_err = snddriver_dsp_write(cmd_port,filtre, Filter_length,sizeof(int),
SNDDRIVER_LOW_PRIORITY);
/////////////// Allocate virtual memory for DMA_IN //////////////////
vm_allocate(task_self(),(vm_address_t *)(&foo),vm_page_size,TRUE);
Error(k_err,"VM Allocation Error ");
vm_allocate(task_self(),(vm_address_t *)(&faa),4*DMASIZE,TRUE);
Error(k_err,"VM Allocation Error ");
beginSound = (short*)aSound + aSound->dataLocation/ sizeof(short) + begin;
endSound = (short*)aSound+aSound->dataLocation/sizeof(short)+begin+length;
beginPage = (((int)beginSound) % (vm_page_size)) / sizeof(short);
endPage = (((int)endSound) % (vm_page_size)) / sizeof(short);
pageNumber = (((int)endSound - (int)beginSound)/sizeof(short)
- endPage - (vm_page_size/2 - beginPage));
if(pageNumber > 0) pageNumber /= (vm_page_size/2);
else pageNumber = 0;
if(INIT) usleep(1500000);
for(i=0;i<length && beginPage<vm_page_size/2;i++, beginPage++)
foo[beginPage] = *(beginSound++);
if(i == length) pageNumber = -1; // No need to send anything else.
k_err = snddriver_stream_start_writing(write_port,(void *)foo,
vm_page_size/2,WRITE_TAG,0,0,0,0,0,0,0,0, reply_port);
if(pageNumber > 0)
{
k_err = snddriver_stream_start_writing(write_port,(void *)beginSound,
pageNumber * vm_page_size/2,WRITE_TAG,0,0,0,0,0,0,0,0, reply_port);
Error(k_err,"Error when starting playing");
}
if(pageNumber >= 0)
{
for(j=0, endSound -= endPage;j<endPage;j++)
foo[j] = *(endSound++);
for(;j<vm_page_size/2;j++)
foo[j] = 0;
k_err = snddriver_stream_start_writing(write_port,(void *)foo,
vm_page_size/2,WRITE_TAG,0,0,0,0,0,0,0,0, reply_port);
Error(k_err,"Error when starting playing");
}
k_err = snddriver_dsp_host_cmd(cmd_port,20,SNDDRIVER_LOW_PRIORITY);
k_err = snddriver_stream_start_writing(write_port,(void *)faa,
2*DMASIZE,WRITE_TAG,0,0,1,0,0,0,0,0, reply_port);
Error(k_err,"Error when finishing playing");
if(WAIT)
{
reply_msg = (msg_header_t *)malloc(MSG_SIZE_MAX);
reply_msg->msg_size = MSG_SIZE_MAX;
reply_msg->msg_local_port = reply_port;
k_err = msg_receive(reply_msg, MSG_OPTION_NONE, 0);
[self playCompleted:self];
return 0;
}
DPSAddPort(reply_port,
HandleDSPMessage, /* function to call */
MSG_SIZE_MAX,
self, /* first arg to HandleDSPMessage */
NX_RUNMODALTHRESHOLD /* priority */
);
play_mon = DPSAddTimedEntry(0.05, (DPSTimedEntryProc) PlayMonitor, self, NX_MODALRESPTHRESHOLD);
status = PLAYING;
return 0;
}
// Calculates FIR low-pass filter by windowing a sinc with a hamming window,
// and normalizing.
- calculateFilter:(int*)afiltre:(int)order:(float)freq_cut:(int)P
{
int i;
float scaler;
float aux;
for(i=1,scaler = 0;i<order/2;i++)
{
aux = (sin(freq_cut * i) / freq_cut / i
* (0.54 + 0.46 * cos(6.28318530*i/(order-2))));
afiltre[order/2+i] = afiltre[order/2-i] = 8388607 * aux ;
scaler += 2 * aux;
}
scaler = P * gain / (1 + scaler);
afiltre[order/2] = 8388607 * scaler;
for(i=1;i<order/2;i++)
afiltre[order/2+i] = afiltre[order/2-i] = afiltre[order/2-i]*scaler;
return self;
}
/* calculateRatio returns in rat a ratio that best approximates the real
x with an relative error less than thresh, making sure U and D are less than
100. Thresh is equal to 0.03, the auditory threshold for pitch sensation.
*/
- calculateRatio:(float)x:(int*)arat:(float)athresh
{
int i=0,j;
float error = 1;
float reste;
int *A, *p, *q;
int prev_p = 1, prev_q = 1;
A = (int*) calloc(20,sizeof(int));
p = (int*) calloc(20,sizeof(int));
q = (int*) calloc(20,sizeof(int));
reste = x;
while(error/x > athresh)
{
A[i] = floor(reste);
reste = 1 / (reste - A[i]);
for(j=i, p[i]=1, q[i]=A[i]; j>0; j--)
{
p[j-1] = q[j];
q[j-1] = A[j-1] * q[j] + p[j];
}
error = fabs((float)q[0]/(float)p[0] - x);
if(p[0] < max_order && q[0] < max_order)
{
prev_p = p[0];
prev_q = q[0];
}
else
{
printf("Warning, difficult conversion: ideal factors U = %d, D = %d,\n",
p[0],q[0]);
p[0] = prev_p;
q[0] = prev_q;
error = fabs((float)q[0]/(float)p[0] - x);
printf("Actual factors: U = %d, D = %d, \
Resulting Sampling rate error: %.2f %%\n",p[0],q[0],100*error/x);
break;
}
i++;
}
arat[0] = q[0];
arat[1] = p[0];
return self;
}
- stop
{
if(status != IDLE)
{
DPSRemovePort(reply_port);
if(play_mon) DPSRemoveTimedEntry(play_mon);
play_mon = 0;
SNDRelease(SND_ACCESS_OUT|SND_ACCESS_DSP,dev_port,owner_port);
vm_deallocate(task_self(),(pointer_t)foo, vm_page_size);
vm_deallocate(task_self(),(pointer_t)faa, 4*DMASIZE);
k_err = port_deallocate(task_self(),reply_port);
free(filtre);
status = IDLE;
if (delegate && [delegate respondsTo:@selector(didPlay:)])
[delegate perform:@selector(didPlay:) with:self];
if (delegate && [delegate respondsTo:@selector(soundAnimate:)])
[delegate perform:@selector(soundAnimate:) with:(id)20];
return self;
}
return self;
}
-(int) pause
{
if(AES != 0 || status == IDLE) return 0;
switch(status)
{
case PLAYING :
k_err = snddriver_dsp_host_cmd(cmd_port,21,SNDDRIVER_LOW_PRIORITY);
Error(k_err,"Couldn't pause");
if(play_mon) DPSRemoveTimedEntry(play_mon);
play_mon = 0;
status = PAUSED; break;
case PAUSED :
k_err = snddriver_dsp_host_cmd(cmd_port,20,SNDDRIVER_LOW_PRIORITY);
Error(k_err,"Couldn't resume");
play_mon = DPSAddTimedEntry(0.05, (DPSTimedEntryProc) PlayMonitor,
self, NX_MODALRESPTHRESHOLD);
status = PLAYING; break;
default : return 0;
}
return 1;
}
- pause:sender
{
[self pause];
return self;
}
- playCompleted:tag
{
usleep(100000);
[self stop];
// if (delegate && [delegate respondsTo:@selector(didPlay:)])
// [delegate perform:@selector(didPlay:) with:self];
return self;
}
- playMonitor
{
int samples;
int isStereo = (stereo+1);
int del = (delegate && [delegate respondsTo:@selector(soundAnimate:)]);
k_err = snddriver_stream_nsamples(write_port, &samples);
samples /= sizeof(short); // samples originally contains nb of bytes!
if (del)
{
if(samples/isStereo > size_samp)
[delegate perform:@selector(soundAnimate:) with:0];
else
[delegate perform:@selector(soundAnimate:)
with:(id)((first_samp+samples)/isStereo)];
}
return self;
}
- setWait:(BOOL)wait
{
WAIT = wait;
return self;
}
- setGain:(float)aGain
{
gain = aGain;
return self;
}
@endThese are the contents of the former NiCE NeXT User Group NeXTSTEP/OpenStep software archive, currently hosted by Netfuture.ch.