Merge pull request #151 from shorepine/partials

Get `partials.py` and `partials.c` up to date with latest AMY, including Loris decomposition playback

Makefile

@@ -7,6 +7,9 @@ LIBS = -lpthread -lm
 # on macOS, need to link to AudioUnit, CoreAudio, and CoreFoundation
 ifeq ($(shell uname -s), Darwin)
 LIBS += -framework AudioUnit -framework CoreAudio -framework CoreFoundation
+
+# Needed for brew's python3.12+ on MacOS
+EXTRA_PIP_ENV = PIP_BREAK_SYSTEM_PACKAGES=1 
 endif
 
 # on Raspberry Pi, at least under 32-bit mode, libatomic and libdl are needed.
@@ -66,7 +69,7 @@ amy-message: $(OBJECTS) src/amy-message.o
  $(CC) $(OBJECTS) src/amy-message.o -Wall $(LIBS) -o $@
 
 amy-module: amy-example
- ${PYTHON} -m pip install -r requirements.txt; touch src/amy.c; cd src; ${PYTHON} -m pip install . --force-reinstall; cd ..
+ ${EXTRA_PIP_ENV} ${PYTHON} -m pip install -r requirements.txt; touch src/amy.c; cd src; ${EXTRA_PIP_ENV} ${PYTHON} -m pip install . --force-reinstall; cd ..
 
 test: amy-module
  ${PYTHON} test.py

README.md

@@ -251,11 +251,13 @@ Here's the full list:
 | S | reset | uint | resets given oscillator. set to > OSCS to reset all oscillators, gain and EQ |
 | s | pitch_bend | float | Sets the global pitch bend, by default modifying all note frequencies by (fractional) octaves up or down |
 | t | time | uint | ms of expected playback since some fixed start point on your host. you should always give this if you can. |
+| T | eg0_type | uint 0-3 | Type for Envelope Generator 0 - 0: Normal (RC-like) / 1: Linear / 2: DX7-style / 3: True exponential. |
 | u | store_patch | number,string | store up to 32 patches in RAM with ID number (1024-1055) and AMY message after a comma. Must be sent alone | 
 | v | osc | uint 0 to OSCS-1 | which oscillator to control | 
 | V | volume | float 0-10 | volume knob for entire synth, default 1.0 | 
 | w | wave | uint 0-11 | waveform: [0=SINE, PULSE, SAW_DOWN, SAW_UP, TRIANGLE, NOISE, KS, PCM, ALGO, PARTIAL, PARTIALS, OFF]. default: 0/SINE |
 | x | eq_l | float | in dB, fc=800Hz amount, -15 to 15. 0 is off. default 0. |
+| X | eg1_type | uint 0-3 | Type for Envelope Generator 1 - 0: Normal (RC-like) / 1: Linear / 2: DX7-style / 3: True exponential. |
 | y | eq_m | float | in dB, fc=2500Hz amount, -15 to 15. 0 is off. default 0. |
 | z | eq_h | float | in dB, fc=7500Hz amount, -15 to 15. 0 is off. default 0. | 
 

amy.py

@@ -122,7 +122,7 @@ def message(osc=0, wave=None, patch=None, note=None, vel=None, amp=None, freq=No
  if(bp0 is not None): m = m +"A%s" % (bp0)
  if(bp1 is not None): m = m +"B%s" % (bp1)
  if(eg0_type is not None): m = m + "T" + str(eg0_type)
- if(eg1_type is not None): m = m + "W" + str(eg1_type)
+ if(eg1_type is not None): m = m + "X" + str(eg1_type)
  if(algo_source is not None): m = m +"O%s" % (algo_source)
  if(chained_osc is not None): m = m + "c" + str(chained_osc)
  if(clone_osc is not None): m = m + "C" + str(clone_osc)

partials.py

@@ -1,11 +1,13 @@
 import amy, sys
 import pydub
+# We keep this path hack around for dan, whose system needs it to get Loris to work. You shouldn't need it.
+sys.path.append('/usr/local/lib/python' + '.'.join(sys.version.split('.', 2)[:2]) + '/site-packages')
 import loris
 import time
 import numpy as np
 from math import pi, log2
-from collections import deque
-
+from collections import deque, defaultdict
+from copy import copy
 
 def list_from_py2_iterator(obj, how_many):
  # Oof, the loris object uses some form of iteration that py3 doesn't like. 
@@ -96,6 +98,7 @@ def sequence(filename, max_len_s = 10, amp_floor=-30, hop_time=0.04, max_oscs=am
  min_q_len = max_oscs
  # Now add in a voice / osc # 
  osc_map = {}
+ osc_free_at = {} # map of osc# -> time to safely free at (6ms after you want)
  osc_q = deque(range(max_oscs)) 
  for i,s in enumerate(time_ordered):
  next_idx = -1
@@ -116,6 +119,14 @@ def sequence(filename, max_len_s = 10, amp_floor=-30, hop_time=0.04, max_oscs=am
  # Start the partials at 0
  s[0] = s[0] - first_time
 
+ # Free oscs if it's time to
+ oscs_to_consider_freeing = copy(list(osc_free_at.keys()))
+ for osc in oscs_to_consider_freeing:
+ if(s[0]>=osc_free_at[osc]):
+ #print("freeing osc %d, it's %d and it waited until %d" % (osc, s[0], osc_free_at[osc]))
+ osc_q.appendleft(osc)
+ del osc_free_at[osc]
+
  if(s[4]>=0): #new partial
  if(len(osc_q)):
  osc_map[s[1]] = osc_q.popleft()
@@ -129,7 +140,11 @@ def sequence(filename, max_len_s = 10, amp_floor=-30, hop_time=0.04, max_oscs=am
  sequence.append(s)
  if(s[4] == -2): # last bp
  # Put the oscillator back
- osc_q.appendleft(osc)
+ # per dan, ONLY put this osc back once at least 12ms (i.e., def 1 AMY frame) has gone by after we get a -2
+ #print("not letting osc %d die until 6ms from now %d = %d" % (s[1], s[0], s[0]+6))
+ osc_free_at[s[1]] = s[0] + 12
+ #osc_q.appendleft(osc)
+
  if(len(osc_q) < min_q_len): min_q_len = len(osc_q)
  print("%d partials and %d breakpoints, max oscs used at once was %d" % (partial_count, len(sequence), max_oscs - min_q_len))
  # Fix sustain_ms
@@ -160,37 +175,54 @@ def play(sequence, osc_offset=0, sustain_ms = -1, sustain_len_ms = 0, time_ratio
  print("Moving sustain_ms from %d to %d" % (sustain_ms, sequence[-1][0]-100))
  sustain_ms = sequence[-1][0] - 100
 
+ # Use a default dict so that values we haven't written yet appear as zeros.
+ time_since_osc_onset = defaultdict(int)
+
  for i,s in enumerate(sequence):
  # Wait for the item in the sequence to be close, so I don't overflow the synthesizers' state
  while(my_start_time + (s[0] / time_ratio) > (amy.millis() - 500)):
  time.sleep(0.01)
 
- # Make envelope strings
- bp0 = "0,1.0,%d,%s,0,0" % (s[5] / time_ratio, amy.trunc(s[6]))
- bp1 = "0,0.0,%d,%s,0,0" % (s[5] / time_ratio, amy.trunc(log2_or_0(s[7])))
+ # Make envelope strings. This is weird because we rewrite the envelopes while the oscillator
+ # is running (and the envelopes are part-evaluated) to allow an unlimited number of segments.
+ # To make it work, each time we change it we have to make a placeholder first interval to get
+ # us to the right time offset after the start of the partial (the time frame used by the
+ # envelope generator).
+ osc = s[1] + osc_offset
+ delta_time = int(round(s[5] / time_ratio))
+ bp0 = "%d,1.0,%d,%s,0,0" % (time_since_osc_onset[osc], delta_time, amy.trunc(s[6]))
+ bp1 = "%d,0.0,%d,%s,0,0" % (time_since_osc_onset[osc], delta_time, amy.trunc(log2_or_0(s[7])))
+ # Update the base time for the next segment.
+ time_since_osc_onset[osc] += delta_time
+
  if(sustain_ms > 0 and sustain_offset == 0):
  if(s[0]/time_ratio > sustain_ms/time_ratio):
  sustain_offset = sustain_len_ms/time_ratio
 
- osc = s[1]+osc_offset
-
- partial_args = {}
-
- partial_args.update({"time":my_start_time + (s[0]/time_ratio + sustain_offset),
- "osc":s[1]+osc_offset,
- "wave":amy.PARTIAL,
+ partial_args = {
+ "time": my_start_time + (s[0]/time_ratio + sustain_offset),
+ "osc": s[1]+osc_offset,
+ "wave": amy.PARTIAL,
  "amp": "%s,0,0,1,0" % amy.trunc(s[3]*amp_ratio),
- "freq":"%s,0,0,0,1" % amy.trunc(s[2]*pitch_ratio),
- "bp0":bp0, "bp1":bp1, "eg0_type": amy.ENVELOPE_LINEAR, "eg1_type":amy.ENVELOPE_LINEAR
- })
+ "freq": "%s,0,0,0,1" % amy.trunc(s[2]*pitch_ratio),
+ "bp0": bp0,
+ "bp1": bp1,
+ "eg0_type": amy.ENVELOPE_LINEAR,
+ "eg1_type": amy.ENVELOPE_LINEAR
+ }
 
  if(s[4]==-2): #end, add note off
- partial_args['amp'] = "0,0,0,1,0"
- amy.send(**partial_args, vel=0)
+ #partial_args['amp'] = "0,0,0,1,0"
+ #partial_args['vel'] = 0
+ # Reset the incremental envelope segment start time.
+ time_since_osc_onset[osc] = 0
  elif(s[4]==-1): # continue
- amy.send(**partial_args)
+ pass
  else: #start, add phase and note on
- amy.send(**partial_args, vel=s[3]*amp_ratio, phase=s[4])
+ partial_args['vel'] = 1.0 # Velocity value is ignored?
+ partial_args['phase'] = s[4]
+
+ amy.send(**partial_args)
 
  return sequence[-1][0]/time_ratio
 

src/amy.c

@@ -467,6 +467,9 @@ void amy_add_event_internal(struct event e, uint16_t base_osc) {
  if(AMY_IS_SET(e.eq_m)) { d.param=EQ_M; d.data = *(uint32_t *)&e.eq_m; add_delta_to_queue(d); }
  if(AMY_IS_SET(e.eq_h)) { d.param=EQ_H; d.data = *(uint32_t *)&e.eq_h; add_delta_to_queue(d); }
 
+ if(AMY_IS_SET(e.eg_type[0])) { d.param=EG0_TYPE; d.data = e.eg_type[0]; add_delta_to_queue(d); }
+ if(AMY_IS_SET(e.eg_type[1])) { d.param=EG1_TYPE; d.data = e.eg_type[1]; add_delta_to_queue(d); }
+
  if(e.algo_source[0] != 0) {
  struct synthinfo t;
  parse_algorithm_source(&t, e.algo_source);
@@ -575,7 +578,7 @@ void reset_osc(uint16_t i ) {
  synth[i].amp_coefs[j] = 0;
  synth[i].amp_coefs[COEF_VEL] = 1.0f;
  synth[i].amp_coefs[COEF_EG0] = 1.0f;
- msynth[i].amp = 1.0f;
+ msynth[i].amp = 0; // This matters for wave=PARTIAL, where msynth amp is effectively 1-frame delayed.
  msynth[i].last_amp = 0;
  for (int j = 0; j < NUM_COMBO_COEFS; ++j)
  synth[i].logfreq_coefs[j] = 0;
@@ -921,6 +924,9 @@ void play_event(struct delta d) {
  synth[synth[d.osc].algo_source[which_source]].status = IS_ALGO_SOURCE;
  }
 
+ if (d.param == EG0_TYPE) synth[d.osc].eg_type[0] = d.data;
+ if (d.param == EG1_TYPE) synth[d.osc].eg_type[1] = d.data;
+
  // for global changes, just make the change, no need to update the per-osc synth
  if(d.param == VOLUME) amy_global.volume = *(float *)&d.data;
  if(d.param == PITCH_BEND) amy_global.pitch_bend = *(float *)&d.data;
@@ -1068,16 +1074,22 @@ void hold_and_modify(uint16_t osc) {
  msynth[osc].pan = combine_controls(ctrl_inputs, synth[osc].pan_coefs);
  // amp is a special case - coeffs apply in log domain.
  // Also, we advance one frame by writing both last_amp and amp (=next amp)
- float new_last_amp = combine_controls_mult(ctrl_inputs, synth[osc].amp_coefs);
- // Prevent hard-off on transition to release by updating last_amp only for nonzero new_last_amp.
- if (new_last_amp > 0) {
- msynth[osc].last_amp = new_last_amp;
+ // *Except* for partials, where we allow one frame of ramp-on.
+ float new_amp = combine_controls_mult(ctrl_inputs, synth[osc].amp_coefs);
+ if (synth[osc].wave == PARTIAL) {
+ msynth[osc].last_amp = msynth[osc].amp;
+ msynth[osc].amp = new_amp;
+ } else {
+ // Prevent hard-off on transition to release by updating last_amp only for nonzero new_last_amp.
+ if (new_amp > 0) {
+ msynth[osc].last_amp = new_amp;
+ }
+ // Advance the envelopes to the beginning of the next frame.
+ ctrl_inputs[COEF_EG0] = S2F(compute_breakpoint_scale(osc, 0, AMY_BLOCK_SIZE));
+ ctrl_inputs[COEF_EG1] = S2F(compute_breakpoint_scale(osc, 1, AMY_BLOCK_SIZE));
+ msynth[osc].amp = combine_controls_mult(ctrl_inputs, synth[osc].amp_coefs);
+ if (msynth[osc].amp <= 0.001) msynth[osc].amp = 0;
  }
- ctrl_inputs[COEF_EG0] = S2F(compute_breakpoint_scale(osc, 0, AMY_BLOCK_SIZE));
- ctrl_inputs[COEF_EG1] = S2F(compute_breakpoint_scale(osc, 1, AMY_BLOCK_SIZE));
- msynth[osc].amp = combine_controls_mult(ctrl_inputs, synth[osc].amp_coefs);
- if (msynth[osc].amp <= 0.001) msynth[osc].amp = 0;
-
  // synth[osc].feedback is copied to msynth in pcm_note_on, then used to track note-off for looping PCM.
  // For PCM, don't re-copy it every loop, or we'd lose track of that flag. (This means you can't change feedback mid-playback for PCM).
  if (synth[osc].wave != PCM) msynth[osc].feedback = synth[osc].feedback;
@@ -1623,14 +1635,14 @@ struct event amy_parse_message(char * message) {
  case 'S': e.reset_osc = atoi(message + start); break;
  case 's': e.pitch_bend = atoff(message + start); break;
  /* t used for time */
- /* T unused */
+ case 'T': e.eg_type[0] = atoi(message + start); break;
  /* U used by Alles for sync */
  case 'u': patches_store_patch(message+start); AMY_PROFILE_STOP(AMY_PARSE_MESSAGE) return amy_default_event(); 
  case 'v': e.osc=((atoi(message + start)) % (AMY_OSCS+1)); break; // allow osc wraparound
  case 'V': e.volume = atoff(message + start); break;
  case 'w': e.wave=atoi(message + start); break;
  /* W used by Tulip for CV, external_channel */
- /* X available */
+ case 'X': e.eg_type[1] = atoi(message + start); break;
  case 'x': e.eq_l = atoff(message+start); break;
  /* Y available */
  case 'y': e.eq_m = atoff(message+start); break;

src/amy.h

@@ -152,9 +152,10 @@ enum params{
  ALGO_SOURCE_END=100+MAX_ALGO_OPS, // 106
  BP_START=ALGO_SOURCE_END + 1, // 107..138
  BP_END=BP_START + (MAX_BREAKPOINT_SETS * MAX_BREAKPOINTS * 2), // 139
- CLONE_OSC, // 140
- RESET_OSC, // 141
- NO_PARAM // 142
+ EG0_TYPE, EG1_TYPE, // 140, 141
+ CLONE_OSC, // 142
+ RESET_OSC, // 143
+ NO_PARAM // 144
 };
 
 #ifdef AMY_DEBUG

src/oscillators.c

@@ -522,7 +522,7 @@ SAMPLE render_partial(SAMPLE * buf, uint16_t osc) {
  PHASOR step = F2P(freq / (float)AMY_SAMPLE_RATE); // cycles per sec / samples per sec -> cycles per sample
  SAMPLE amp = F2S(msynth[osc].amp);
  SAMPLE last_amp = F2S(msynth[osc].last_amp);
- //printf("render_partial: time %lld logfreq %f freq %f amp %f step %f\n", total_samples, msynth[osc].logfreq, freq, S2F(amp), P2F(step) * synth[osc].lut->table_size); 
+ //printf("render_partial: time %.3f logfreq %f freq %f last_amp %f amp %f step %f\n", (float)total_samples/(float)AMY_SAMPLE_RATE, msynth[osc].logfreq, freq, S2F(last_amp), S2F(amp), P2F(step) * synth[osc].lut->table_size); 
  SAMPLE max_value;
  synth[osc].phase = render_lut(buf, synth[osc].phase, step, last_amp, amp, synth[osc].lut, &max_value);
  msynth[osc].last_amp = msynth[osc].amp;
@@ -533,6 +533,8 @@ void partial_note_on(uint16_t osc) {
  float freq = freq_of_logfreq(msynth[osc].logfreq);
  float period_samples = (float)AMY_SAMPLE_RATE / freq;
  synth[osc].lut = choose_from_lutset(period_samples, sine_fxpt_lutset);
+ // Partials ramp up from zero.
+ msynth[osc].amp = 0;
 }
 
 void partial_note_off(uint16_t osc) {

src/partials.c

@@ -71,7 +71,7 @@ SAMPLE render_partials(SAMPLE *buf, uint16_t osc) {
  // Find our ratio using the midi note of the analyzed partial
  float freq_logratio = msynth[osc].logfreq - logfreq_for_midi_note(patch.midi_note);
 
- //printf("time %f rel %f: freqlogratio %f new pb: osc %d t_ms %d amp %f freq %f phase %f logfreq %f\n", total_samples / (float)AMY_SAMPLE_RATE, ms_since_started / (float)AMY_SAMPLE_RATE, freq_logratio, pb.osc, pb.ms_offset, pb.amp, pb.freq, pb.phase, logfreq_of_freq(pb.freq));
+ //printf("time %.3f rel %f: freqlogratio %f new pb: osc %d t_ms %d amp %f freq %f phase %f logfreq %f\n", total_samples / (float)AMY_SAMPLE_RATE, ms_since_started / (float)AMY_SAMPLE_RATE, freq_logratio, pb.osc, pb.ms_offset, pb.amp, pb.freq, pb.phase, logfreq_of_freq(pb.freq));
 
  // All the types share these params or are overwritten
  synth[o].wave = PARTIAL;
@@ -116,11 +116,10 @@ SAMPLE render_partials(SAMPLE *buf, uint16_t osc) {
  synth[o].logfreq_coefs[0] = logfreq_of_freq(pb.freq) + freq_logratio;
  //printf("[%d %d] o %d continue partial\n", total_samples, ms_since_started, o);
  } else if(partial_code==2) { // partial is done, give it one buffer to ramp to zero.
- synth[o].amp_coefs[0] = 0;
+ synth[o].amp_coefs[0] = 0.0001;
  //partial_note_off(o);
  } else { // start of a partial, 
  //printf("[%d %d] o %d start partial\n", total_samples,ms_since_started, o);
- msynth[o].last_amp = 0;
  partial_note_on(o);
  }
  synth[osc].step++;
@@ -137,12 +136,10 @@ SAMPLE render_partials(SAMPLE *buf, uint16_t osc) {
  for(uint16_t i=osc+1;i<osc+1+oscs;i++) {
  uint16_t o = i % AMY_OSCS;
  if(synth[o].status ==IS_ALGO_SOURCE) {
- // msynth amp used to lag one frame behind, but we advanced it one frame.
- // For partials, retard it again to preserve the old behavior.
- float last_amp_on_entry = msynth[o].last_amp;
+ // hold_and_modify contains a special case for wave == PARTIAL so that
+ // envelope value are delayed by 1 frame compared to other oscs
+ // so that partials fade in over one frame from zero amp.
  hold_and_modify(o);
- msynth[o].amp = msynth[o].last_amp;
- msynth[o].last_amp = last_amp_on_entry;
  //printf("[%d %d] %d amp %f (%f) freq %f (%f) on %d off %d bp0 %d %f bp1 %d %f wave %d\n", total_samples, ms_since_started, o, synth[o].amp, msynth[o].amp, synth[o].freq, msynth[o].freq, synth[o].note_on_clock, synth[o].note_off_clock, synth[o].breakpoint_times[0][0], 
  // synth[o].breakpoint_values[0][0], synth[o].breakpoint_times[1][0], synth[o].breakpoint_values[1][0], synth[o].wave);
  //for(uint16_t j=0;j<AMY_BLOCK_SIZE;j++) pbuf[j] = 0;