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authorStepan Moskovchenko <stevenm@rockbox.org>2007-10-15 05:11:37 +0000
committerStepan Moskovchenko <stevenm@rockbox.org>2007-10-15 05:11:37 +0000
commit1515ff852224c822a6d3db8c458eab2c9037704f (patch)
treee427fbec1b397d18abffc12b7fe74e67c2cad807
parent99f955088149d5938ce4c9ca5624377f464b1380 (diff)
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MIDI: At long last, though quick and dirty, pitch bend depth! Or, I think it works. Tested on two
files. Let me know if anyone discovers any problems with this. This commit also includes Nils's synth loop optimization patch. I hope committing it does not cause problems. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@15112 a1c6a512-1295-4272-9138-f99709370657
-rw-r--r--apps/plugins/midi/midiplay.c14
-rw-r--r--apps/plugins/midi/midiutil.c1
-rw-r--r--apps/plugins/midi/midiutil.h4
-rw-r--r--apps/plugins/midi/sequencer.c78
-rw-r--r--apps/plugins/midi/synth.c261
5 files changed, 218 insertions, 140 deletions
diff --git a/apps/plugins/midi/midiplay.c b/apps/plugins/midi/midiplay.c
index 99f05718d6..325d90c375 100644
--- a/apps/plugins/midi/midiplay.c
+++ b/apps/plugins/midi/midiplay.c
@@ -93,6 +93,7 @@ int numberOfSamples IBSS_ATTR;
long bpm IBSS_ATTR;
int32_t gmbuf[BUF_SIZE*NBUF];
+static unsigned int samples_in_buf;
int quit=0;
struct plugin_api * rb;
@@ -160,7 +161,8 @@ static inline void synthbuf(void)
outptr=gmbuf;
#endif
- for(i=0; i<BUF_SIZE/numberOfSamples; i++)
+ /* synth samples for as many whole ticks as we can fit in the buffer */
+ for(i=0; i < BUF_SIZE/numberOfSamples; i++)
{
synthSamples((int32_t*)outptr, numberOfSamples);
outptr += numberOfSamples;
@@ -168,11 +170,9 @@ static inline void synthbuf(void)
quit=1;
}
- if(BUF_SIZE%numberOfSamples)
- {
- synthSamples((int32_t*)outptr, BUF_SIZE%numberOfSamples);
- outptr += BUF_SIZE%numberOfSamples;
- }
+ /* how many samples did we write to the buffer? */
+ samples_in_buf = BUF_SIZE-(BUF_SIZE%numberOfSamples);
+
}
void get_more(unsigned char** start, size_t* size)
@@ -187,7 +187,7 @@ void get_more(unsigned char** start, size_t* size)
synthbuf(); // For some reason midiplayer crashes when an update is forced
#endif
- *size = sizeof(gmbuf)/NBUF;
+ *size = samples_in_buf*sizeof(int32_t);
#ifndef SYNC
*start = (unsigned char*)((swap ? gmbuf : gmbuf + BUF_SIZE));
swap=!swap;
diff --git a/apps/plugins/midi/midiutil.c b/apps/plugins/midi/midiutil.c
index aba56c5a8c..8cf8ffcde6 100644
--- a/apps/plugins/midi/midiutil.c
+++ b/apps/plugins/midi/midiutil.c
@@ -25,6 +25,7 @@ int chVol[16] IBSS_ATTR; /* Channel volume */
int chPan[16] IBSS_ATTR; /* Channel panning */
int chPat[16] IBSS_ATTR; /* Channel patch */
int chPW[16] IBSS_ATTR; /* Channel pitch wheel, MSB only */
+int chPBDepth[16] IBSS_ATTR; /* Channel pitch wheel, MSB only */
struct GPatch * gusload(char *);
struct GPatch * patchSet[128];
diff --git a/apps/plugins/midi/midiutil.h b/apps/plugins/midi/midiutil.h
index a94c257df0..911774440e 100644
--- a/apps/plugins/midi/midiutil.h
+++ b/apps/plugins/midi/midiutil.h
@@ -63,7 +63,8 @@
#define MIDI_PITCHW 224
/* MIDI Controllers */
-#define CTRL_VOLUME 7
+#define CTRL_PWDEPTH 6
+#define CTRL_VOLUME 7
#define CTRL_BALANCE 8
#define CTRL_PANNING 10
#define CHANNEL 1
@@ -159,6 +160,7 @@ extern int chVol[16]; /* Channel volume */
extern int chPan[16]; /* Channel panning */
extern int chPat[16]; /* Channel patch */
extern int chPW[16]; /* Channel pitch wheel, MSB only */
+extern int chPBDepth[16]; /* Channel pitch bend depth (Controller 6 */
extern struct GPatch * gusload(char *);
extern struct GPatch * patchSet[128];
diff --git a/apps/plugins/midi/sequencer.c b/apps/plugins/midi/sequencer.c
index 1a00c078c6..638c9ba43a 100644
--- a/apps/plugins/midi/sequencer.c
+++ b/apps/plugins/midi/sequencer.c
@@ -75,7 +75,60 @@ long pitchTbl[]=
};
*/
+
+/* 512 entries here */
+/*
+ for i=0:512, fprintf('%d,', round(2^16*2^((i-256)/1536))); end
+*/
+
const uint32_t pitchTbl[] ICONST_ATTR={
+ 61858,61872,61886,61900,61914,61928,61942,61956,61970,61983,61997,62011,
+ 62025,62039,62053,62067,62081,62095,62109,62124,62138,62152,62166,62180,
+ 62194,62208,62222,62236,62250,62264,62278,62292,62306,62320,62334,62348,
+ 62362,62376,62390,62404,62419,62433,62447,62461,62475,62489,62503,62517,
+ 62531,62545,62560,62574,62588,62602,62616,62630,62644,62658,62673,62687,
+ 62701,62715,62729,62743,62757,62772,62786,62800,62814,62828,62843,62857,
+ 62871,62885,62899,62913,62928,62942,62956,62970,62984,62999,63013,63027,
+ 63041,63056,63070,63084,63098,63112,63127,63141,63155,63169,63184,63198,
+ 63212,63227,63241,63255,63269,63284,63298,63312,63326,63341,63355,63369,
+ 63384,63398,63412,63427,63441,63455,63470,63484,63498,63512,63527,63541,
+ 63555,63570,63584,63599,63613,63627,63642,63656,63670,63685,63699,63713,
+ 63728,63742,63757,63771,63785,63800,63814,63829,63843,63857,63872,63886,
+ 63901,63915,63929,63944,63958,63973,63987,64002,64016,64030,64045,64059,
+ 64074,64088,64103,64117,64132,64146,64161,64175,64190,64204,64219,64233,
+ 64248,64262,64277,64291,64306,64320,64335,64349,64364,64378,64393,64407,
+ 64422,64436,64451,64465,64480,64494,64509,64524,64538,64553,64567,64582,
+ 64596,64611,64626,64640,64655,64669,64684,64699,64713,64728,64742,64757,
+ 64772,64786,64801,64815,64830,64845,64859,64874,64889,64903,64918,64933,
+ 64947,64962,64976,64991,65006,65020,65035,65050,65065,65079,65094,65109,
+ 65123,65138,65153,65167,65182,65197,65211,65226,65241,65256,65270,65285,
+ 65300,65315,65329,65344,65359,65374,65388,65403,65418,65433,65447,65462,
+ 65477,65492,65506,65521,65536,65551,65566,65580,65595,65610,65625,65640,
+ 65654,65669,65684,65699,65714,65729,65743,65758,65773,65788,65803,65818,
+ 65832,65847,65862,65877,65892,65907,65922,65936,65951,65966,65981,65996,
+ 66011,66026,66041,66056,66071,66085,66100,66115,66130,66145,66160,66175,
+ 66190,66205,66220,66235,66250,66265,66280,66294,66309,66324,66339,66354,
+ 66369,66384,66399,66414,66429,66444,66459,66474,66489,66504,66519,66534,
+ 66549,66564,66579,66594,66609,66624,66639,66654,66670,66685,66700,66715,
+ 66730,66745,66760,66775,66790,66805,66820,66835,66850,66865,66880,66896,
+ 66911,66926,66941,66956,66971,66986,67001,67016,67032,67047,67062,67077,
+ 67092,67107,67122,67137,67153,67168,67183,67198,67213,67228,67244,67259,
+ 67274,67289,67304,67320,67335,67350,67365,67380,67395,67411,67426,67441,
+ 67456,67472,67487,67502,67517,67532,67548,67563,67578,67593,67609,67624,
+ 67639,67655,67670,67685,67700,67716,67731,67746,67761,67777,67792,67807,
+ 67823,67838,67853,67869,67884,67899,67915,67930,67945,67961,67976,67991,
+ 68007,68022,68037,68053,68068,68083,68099,68114,68129,68145,68160,68176,
+ 68191,68206,68222,68237,68252,68268,68283,68299,68314,68330,68345,68360,
+ 68376,68391,68407,68422,68438,68453,68468,68484,68499,68515,68530,68546,
+ 68561,68577,68592,68608,68623,68639,68654,68670,68685,68701,68716,68732,
+ 68747,68763,68778,68794,68809,68825,68840,68856,68871,68887,68902,68918,
+ 68933,68949,68965,68980,68996,69011,69027,69042,69058,69074,69089,69105,
+ 69120,69136,69152,69167,69183,69198,69214,69230,69245,69261,69276,69292,
+ 69308,69323,69339,69355,69370,69386,69402,69417,69433
+
+};
+
+/*
58386,58412,58439,58465,58491,58518,58544,58571,58597,58624,58650,58676,
58703,58729,58756,58782,58809,58836,58862,58889,58915,58942,58968,58995,
59022,59048,59075,59102,59128,59155,59182,59208,59235,59262,59289,59315,
@@ -119,7 +172,10 @@ const uint32_t pitchTbl[] ICONST_ATTR={
72507,72540,72573,72605,72638,72671,72704,72736,72769,72802,72835,72868,
72901,72934,72967,72999,73032,73065,73098,73131,73164,73197,73230,73264,
73297,73330,73363,73396,73429,73462,73495,73528
-};
+};*/
+
+
+
static void findDelta(struct SynthObject * so, int ch, int note)
{
@@ -128,8 +184,15 @@ static void findDelta(struct SynthObject * so, int ch, int note)
so->wf=wf;
unsigned int delta= 0;
- delta = (((gustable[note]<<FRACTSIZE) / (wf->rootFreq)) * wf->sampRate / (SAMPLE_RATE));
- delta = (delta * pitchTbl[chPW[ch]])>> 16;
+ int totalBend = (chPW[ch]-256) * chPBDepth[ch];
+
+ int noteOffset = totalBend >> 8;
+
+ int pitchOffset = totalBend - (noteOffset<<8);
+
+
+ delta = (((gustable[note+noteOffset]<<FRACTSIZE) / (wf->rootFreq)) * wf->sampRate / (SAMPLE_RATE));
+ delta = (delta * pitchTbl[pitchOffset+256])>> 16;
so->delta = delta;
}
@@ -280,6 +343,13 @@ static void sendEvent(struct Event * ev)
chPan[status_low]=d2;
return;
}
+ case CTRL_PWDEPTH:
+ {
+ /* TODO: Update all deltas. Is this really needed? */
+ chPBDepth[status_low] = d2;
+ return;
+ }
+
}
break;
@@ -293,7 +363,7 @@ static void sendEvent(struct Event * ev)
case 0: /* Release by vol=0 */
releaseNote(status_low, d1);
return;
-
+
default:
pressNote(status_low, d1, d2);
return;
diff --git a/apps/plugins/midi/synth.c b/apps/plugins/midi/synth.c
index 568c7bb1ce..f0fa93d60e 100644
--- a/apps/plugins/midi/synth.c
+++ b/apps/plugins/midi/synth.c
@@ -65,6 +65,7 @@ int initSynth(struct MIDIfile * mf, char * filename, char * drumConfig)
chPan[a]=64; /* Center */
chPat[a]=0; /* Ac Gr Piano */
chPW[a]=256; /* .. not .. bent ? */
+ chPBDepth[a]=2; /* Default bend value is 2 */
}
for(a=0; a<128; a++)
{
@@ -255,191 +256,195 @@ inline void stopVoice(struct SynthObject * so)
so->decay = 0;
}
-static inline int synthVoice(struct SynthObject * so)
+static inline void synthVoice(struct SynthObject * so, int32_t * out, unsigned int samples)
{
struct GWaveform * wf;
register int s;
- register unsigned int cpShifted;
- register short s1;
- register short s2;
+ register int s1;
+ register int s2;
+
+ register unsigned int cp_temp = so->cp;
wf = so->wf;
+ const int mode_mask24 = wf->mode&24;
+ const int mode_mask28 = wf->mode&28;
+ const int mode_mask_looprev = wf->mode&LOOP_REVERSE;
- /* Is voice being ramped? */
- if(so->state == STATE_RAMPDOWN)
- {
- if(so->decay != 0) /* Ramp has been started */
- {
- so->decay = so->decay / 2;
+ const unsigned int num_samples = (wf->numSamples-1) << FRACTSIZE;
- if(so->decay < 10 && so->decay > -10)
- so->isUsed = 0;
+ const unsigned int end_loop = wf->endLoop << FRACTSIZE;
+ const unsigned int start_loop = wf->startLoop << FRACTSIZE;
+ const int diff_loop = end_loop-start_loop;
- return so->decay;
- }
- } else /* OK to advance voice */
+ while(samples > 0)
{
- so->cp += so->delta;
- }
-
-
- cpShifted = so->cp >> FRACTSIZE;
+ samples--;
+ /* Is voice being ramped? */
+ if(so->state == STATE_RAMPDOWN)
+ {
+ if(so->decay != 0) /* Ramp has been started */
+ {
+ so->decay = so->decay / 2;
+ if(so->decay < 10 && so->decay > -10)
+ so->isUsed = 0;
+ s1=so->decay;
+ s2 = s1*chPan[so->ch];
+ s1 = (s1<<7) -s2;
+ *(out++)+=(((s1&0x7FFF80) << 9) | ((s2&0x7FFF80) >> 7));
+ continue;
+ }
+ } else /* OK to advance voice */
+ {
+ cp_temp += so->delta;
+ }
- s2 = getSample((cpShifted)+1, wf);
+ s2 = getSample((cp_temp >> FRACTSIZE)+1, wf);
/* LOOP_REVERSE|LOOP_PINGPONG = 24 */
- if((wf->mode & (24)) && so->loopState == STATE_LOOPING && (cpShifted < (wf->startLoop)))
- {
- if(wf->mode & LOOP_REVERSE)
- {
- cpShifted = wf->endLoop-(wf->startLoop-cpShifted);
- so->cp = (cpShifted)<<FRACTSIZE;
- s2=getSample((cpShifted), wf);
- }
- else
+ if(mode_mask24 && so->loopState == STATE_LOOPING && (cp_temp < start_loop))
{
- so->delta = -so->delta; /* At this point cpShifted is wrong. We need to take a step */
- so->loopDir = LOOPDIR_FORWARD;
+ if(mode_mask_looprev)
+ {
+ cp_temp += diff_loop;
+ s2=getSample((cp_temp >> FRACTSIZE), wf);
+ }
+ else
+ {
+ so->delta = -so->delta; /* At this point cp_temp is wrong. We need to take a step */
+ so->loopDir = LOOPDIR_FORWARD;
+ }
}
- }
- if((wf->mode & 28) && (cpShifted >= wf->endLoop))
- {
- so->loopState = STATE_LOOPING;
- if((wf->mode & (24)) == 0)
+ if(mode_mask28 && (cp_temp >= end_loop))
{
- cpShifted = wf->startLoop + (cpShifted-wf->endLoop);
- so->cp = (cpShifted)<<FRACTSIZE;
- s2=getSample((cpShifted), wf);
+ so->loopState = STATE_LOOPING;
+ if(!mode_mask24)
+ {
+ cp_temp -= diff_loop;
+ s2=getSample((cp_temp >> FRACTSIZE), wf);
+ }
+ else
+ {
+ so->delta = -so->delta;
+ so->loopDir = LOOPDIR_REVERSE;
+ }
}
- else
+
+ /* Have we overrun? */
+ if(cp_temp >= num_samples)
{
- so->delta = -so->delta;
- so->loopDir = LOOPDIR_REVERSE;
+ cp_temp -= so->delta;
+ s2 = getSample((cp_temp >> FRACTSIZE)+1, wf);
+ stopVoice(so);
}
- }
-
- /* Have we overrun? */
- if( (cpShifted >= (wf->numSamples-1)))
- {
- so->cp -= so->delta;
- cpShifted = so->cp >> FRACTSIZE;
- s2 = getSample((cpShifted)+1, wf);
- stopVoice(so);
- }
-
-
- /* Better, working, linear interpolation */
- s1=getSample((cpShifted), wf);
- s = s1 + ((signed)((s2 - s1) * (so->cp & ((1<<FRACTSIZE)-1)))>>FRACTSIZE);
+ /* Better, working, linear interpolation */
+ s1=getSample((cp_temp >> FRACTSIZE), wf);
+ s = s1 + ((signed)((s2 - s1) * (cp_temp & ((1<<FRACTSIZE)-1)))>>FRACTSIZE);
- if(so->curRate == 0)
- {
- stopVoice(so);
-// so->isUsed = 0;
+ if(so->curRate == 0)
+ {
+ stopVoice(so);
+// so->isUsed = 0;
- }
+ }
- if(so->ch != 9 && so->state != STATE_RAMPDOWN) /* Stupid ADSR code... and don't do ADSR for drums */
- {
- if(so->curOffset < so->targetOffset)
+ if(so->ch != 9 && so->state != STATE_RAMPDOWN) /* Stupid ADSR code... and don't do ADSR for drums */
{
- so->curOffset += (so->curRate);
- if(so -> curOffset > so->targetOffset && so->curPoint != 2)
+ if(so->curOffset < so->targetOffset)
{
- if(so->curPoint != 5)
+ so->curOffset += (so->curRate);
+ if(so -> curOffset > so->targetOffset && so->curPoint != 2)
{
- setPoint(so, so->curPoint+1);
+ if(so->curPoint != 5)
+ {
+ setPoint(so, so->curPoint+1);
+ }
+ else
+ {
+ stopVoice(so);
+ }
}
- else
+ } else
+ {
+ so->curOffset -= (so->curRate);
+ if(so -> curOffset < so->targetOffset && so->curPoint != 2)
{
- stopVoice(so);
+
+ if(so->curPoint != 5)
+ {
+ setPoint(so, so->curPoint+1);
+ }
+ else
+ {
+ stopVoice(so);
+ }
+
}
}
- } else
+ }
+
+ if(so->curOffset < 0)
{
- so->curOffset -= (so->curRate);
- if(so -> curOffset < so->targetOffset && so->curPoint != 2)
- {
+ so->curOffset = so->targetOffset;
+ stopVoice(so);
+ }
- if(so->curPoint != 5)
- {
- setPoint(so, so->curPoint+1);
- }
- else
- {
- stopVoice(so);
- }
+ s = (s * (so->curOffset >> 22) >> 8);
- }
+ /* need to set ramp beginning */
+ if(so->state == STATE_RAMPDOWN && so->decay == 0)
+ {
+ so->decay = s*so->volscale>>14;
+ if(so->decay == 0)
+ so->decay = 1; /* stupid junk.. */
}
- }
- if(so->curOffset < 0)
- {
- so->curOffset = so->targetOffset;
- stopVoice(so);
- }
- s = (s * (so->curOffset >> 22) >> 8);
+ /* Scaling by channel volume and note volume is done in sequencer.c */
+ /* That saves us some multiplication and pointer operations */
+ s1=s*so->volscale>>14;
+ s2 = s1*chPan[so->ch];
+ s1 = (s1<<7) - s2;
+ *(out++)+=(((s1&0x7FFF80) << 9) | ((s2&0x7FFF80) >> 7));
- /* need to set ramp beginning */
- if(so->state == STATE_RAMPDOWN && so->decay == 0)
- {
- so->decay = s*so->volscale>>14;
- if(so->decay == 0)
- so->decay = 1; /* stupid junk.. */
}
-
- /* Scaling by channel volume and note volume is done in sequencer.c */
- /* That saves us some multiplication and pointer operations */
- return s*so->volscale>>14;
+ so->cp=cp_temp; /* store this again */
+ return;
}
+/* buffer to hold all the samples for the current tick, this is a hack
+ neccesary for coldfire targets as pcm_play_data uses the dma which cannot
+ access iram */
+int32_t samp_buf[256] IBSS_ATTR;
+
/* synth num_samples samples and write them to the */
/* buffer pointed to by buf_ptr */
void synthSamples(int32_t *buf_ptr, unsigned int num_samples) ICODE_ATTR;
void synthSamples(int32_t *buf_ptr, unsigned int num_samples)
{
int i;
- register int dL;
- register int dR;
- register int sample;
- register struct SynthObject *voicept;
- while(num_samples>0)
- {
- dL=0;
- dR=0;
- voicept=&voices[0];
+ struct SynthObject *voicept;
+
+ rb->memset(samp_buf, 0, num_samples*4);
- for(i=MAX_VOICES; i > 0; i--)
+ for(i=0; i < MAX_VOICES; i++)
+ {
+ voicept=&voices[i];
+ if(voicept->isUsed==1)
{
- if(voicept->isUsed==1)
- {
- sample = synthVoice(voicept);
- dL += sample;
- sample *= chPan[voicept->ch];
- dR += sample;
- }
- voicept++;
+ synthVoice(voicept, samp_buf, num_samples);
}
+ }
- dL = (dL << 7) - dR;
-
- /* combine the left and right 16 bit samples into 32 bits and write */
- /* to the buffer, left sample in the high word and right in the low word */
- *buf_ptr=(((dL&0x7FFF80) << 9) | ((dR&0x7FFF80) >> 7));
+ rb->memcpy(buf_ptr, samp_buf, num_samples*4);
- buf_ptr++;
- num_samples--;
- }
/* TODO: Automatic Gain Control, anyone? */
/* Or, should this be implemented on the DSP's output volume instead? */