static bool Delay_pstate_set_audio_rate(Device_state* dstate, int32_t audio_rate)
{
assert(dstate != NULL);
assert(audio_rate > 0);
Delay_pstate* dpstate = (Delay_pstate*)dstate;
const Proc_delay* delay = (const Proc_delay*)dstate->device->dimpl;
const int32_t delay_buf_size = delay->max_delay * audio_rate + 1;
for (int i = 0; i < KQT_BUFFERS_MAX; ++i)
{
Work_buffer* buf = dpstate->bufs[i];
if (!Work_buffer_resize(buf, delay_buf_size))
return false;
Work_buffer_clear(buf, 0, Work_buffer_get_size(buf));
}
dpstate->buf_pos = 0;
return true;
}
bool Delay_pstate_set_max_delay(
Device_state* dstate, const Key_indices indices, double value)
{
assert(dstate != NULL);
ignore(indices);
ignore(value);
Delay_pstate* dpstate = (Delay_pstate*)dstate;
const Proc_delay* delay = (const Proc_delay*)dstate->device->dimpl;
const int32_t delay_buf_size = delay->max_delay * dstate->audio_rate + 1;
for (int i = 0; i < KQT_BUFFERS_MAX; ++i)
{
Work_buffer* buf = dpstate->bufs[i];
if (!Work_buffer_resize(buf, delay_buf_size))
return false;
Work_buffer_clear(buf, 0, Work_buffer_get_size(buf));
}
dpstate->buf_pos = 0;
return true;
}
static void Device_thread_state_clear_buffers(
Device_thread_state* ts,
Device_buffer_type buf_type,
int32_t buf_start,
int32_t buf_stop)
{
rassert(ts != NULL);
rassert(buf_type < DEVICE_BUFFER_TYPES);
rassert(buf_start >= 0);
rassert(buf_stop >= buf_start);
for (Device_port_type port_type = DEVICE_PORT_TYPE_RECV;
port_type < DEVICE_PORT_TYPES; ++port_type)
{
Etable* bufs = ts->buffers[buf_type][port_type];
const int cap = Etable_get_capacity(bufs);
for (int port = 0; port < cap; ++port)
{
Work_buffer* buffer = Etable_get(bufs, port);
if (buffer != NULL)
Work_buffer_clear(buffer, buf_start, buf_stop);
}
}
Bit_array_clear(ts->in_connected);
return;
}
static void Delay_pstate_reset(Device_state* dstate)
{
assert(dstate != NULL);
Delay_pstate* dpstate = (Delay_pstate*)dstate;
for (int i = 0; i < KQT_BUFFERS_MAX; ++i)
{
Work_buffer* buf = dpstate->bufs[i];
Work_buffer_clear(buf, 0, Work_buffer_get_size(buf));
}
dpstate->buf_pos = 0;
return;
}
static void Delay_pstate_clear_history(Proc_state* proc_state)
{
assert(proc_state != NULL);
Delay_pstate* dpstate = (Delay_pstate*)proc_state;
for (int i = 0; i < KQT_BUFFERS_MAX; ++i)
{
Work_buffer* buf = dpstate->bufs[i];
Work_buffer_clear(buf, 0, Work_buffer_get_size(buf));
}
dpstate->buf_pos = 0;
return;
}
static int32_t Add_vstate_render_voice(
Voice_state* vstate,
Proc_state* proc_state,
const Device_thread_state* proc_ts,
const Au_state* au_state,
const Work_buffers* wbs,
int32_t buf_start,
int32_t buf_stop,
double tempo)
{
rassert(vstate != NULL);
rassert(proc_state != NULL);
rassert(proc_ts != NULL);
rassert(au_state != NULL);
rassert(wbs != NULL);
rassert(tempo > 0);
const Device_state* dstate = &proc_state->parent;
const Proc_add* add = (Proc_add*)proc_state->parent.device->dimpl;
Add_vstate* add_state = (Add_vstate*)vstate;
rassert(is_p2(ADD_BASE_FUNC_SIZE));
// Get frequencies
Work_buffer* freqs_wb = Device_thread_state_get_voice_buffer(
proc_ts, DEVICE_PORT_TYPE_RECV, PORT_IN_PITCH);
Work_buffer* pitches_wb = freqs_wb;
if (freqs_wb == NULL)
freqs_wb = Work_buffers_get_buffer_mut(wbs, ADD_WORK_BUFFER_FIXED_PITCH);
Proc_fill_freq_buffer(freqs_wb, pitches_wb, buf_start, buf_stop);
const float* freqs = Work_buffer_get_contents(freqs_wb);
// Get volume scales
Work_buffer* scales_wb = Device_thread_state_get_voice_buffer(
proc_ts, DEVICE_PORT_TYPE_RECV, PORT_IN_FORCE);
Work_buffer* dBs_wb = scales_wb;
if (scales_wb == NULL)
scales_wb = Work_buffers_get_buffer_mut(wbs, ADD_WORK_BUFFER_FIXED_FORCE);
Proc_fill_scale_buffer(scales_wb, dBs_wb, buf_start, buf_stop);
const float* scales = Work_buffer_get_contents(scales_wb);
// Get output buffer for writing
float* out_bufs[2] = { NULL };
Proc_state_get_voice_audio_out_buffers(
proc_ts, PORT_OUT_AUDIO_L, PORT_OUT_COUNT, out_bufs);
// Get phase modulation signal
const Work_buffer* mod_wbs[] =
{
Device_thread_state_get_voice_buffer(
proc_ts, DEVICE_PORT_TYPE_RECV, PORT_IN_PHASE_MOD_L),
Device_thread_state_get_voice_buffer(
proc_ts, DEVICE_PORT_TYPE_RECV, PORT_IN_PHASE_MOD_R),
};
for (int ch = 0; ch < 2; ++ch)
{
if (mod_wbs[ch] == NULL)
{
Work_buffer* zero_buf = Work_buffers_get_buffer_mut(
wbs, (Work_buffer_type)(ADD_WORK_BUFFER_MOD_L + ch));
Work_buffer_clear(zero_buf, buf_start, buf_stop);
mod_wbs[ch] = zero_buf;
}
}
// Add base waveform tones
const double inv_audio_rate = 1.0 / dstate->audio_rate;
const float* base = Sample_get_buffer(add->base, 0);
for (int h = 0; h < add_state->tone_limit; ++h)
{
const Add_tone* tone = &add->tones[h];
const double pitch_factor = tone->pitch_factor;
const double volume_factor = tone->volume_factor;
if ((pitch_factor <= 0) || (volume_factor <= 0))
continue;
const double pannings[] =
{
-tone->panning,
tone->panning,
};
const double pitch_factor_inv_audio_rate = pitch_factor * inv_audio_rate;
Add_tone_state* tone_state = &add_state->tones[h];
for (int32_t ch = 0; ch < 2; ++ch)
{
float* out_buf_ch = out_bufs[ch];
if (out_buf_ch == NULL)
continue;
const double panning_factor = 1 + pannings[ch];
const float* mod_values_ch = Work_buffer_get_contents(mod_wbs[ch]);
double phase = tone_state->phase[ch];
int32_t res_slice_start = buf_start;
while (res_slice_start < buf_stop)
{
int32_t res_slice_stop = buf_stop;
// Get current pitch range
const float first_mod_shift =
mod_values_ch[res_slice_start] - add_state->prev_mod[ch];
const float first_phase_shift_abs = (float)fabs(
first_mod_shift +
(freqs[res_slice_start] * pitch_factor_inv_audio_rate));
int shift_exp = 0;
const float shift_norm = frexpf(first_phase_shift_abs, &shift_exp);
const float min_phase_shift_abs = ldexpf(0.5f, shift_exp);
const float max_phase_shift_abs = min_phase_shift_abs * 2.0f;
// Choose appropriate waveform resolution for current pitch range
int32_t cur_size = ADD_BASE_FUNC_SIZE;
if (isfinite(shift_norm) && (shift_norm > 0.0f))
{
cur_size = (int32_t)ipowi(2, max(-shift_exp + 1, 3));
cur_size = min(cur_size, ADD_BASE_FUNC_SIZE * 2);
rassert(is_p2(cur_size));
}
const uint32_t cur_size_mask = (uint32_t)cur_size - 1;
const int base_offset = (ADD_BASE_FUNC_SIZE * 4 - cur_size * 2);
rassert(base_offset >= 0);
rassert(base_offset < (ADD_BASE_FUNC_SIZE * 4) - 1);
const float* cur_base = base + base_offset;
// Get length of input compatible with current waveform resolution
const int32_t res_check_stop = min(res_slice_stop,
max(Work_buffer_get_const_start(freqs_wb),
Work_buffer_get_const_start(mod_wbs[ch])) + 1);
for (int32_t i = res_slice_start + 1; i < res_check_stop; ++i)
{
const float cur_mod_shift = mod_values_ch[i] - mod_values_ch[i - 1];
const float cur_phase_shift_abs = (float)fabs(
cur_mod_shift +
(freqs[i] * pitch_factor_inv_audio_rate));
if (cur_phase_shift_abs < min_phase_shift_abs ||
cur_phase_shift_abs > max_phase_shift_abs)
{
res_slice_stop = i;
break;
}
}
for (int32_t i = res_slice_start; i < res_slice_stop; ++i)
{
const float freq = freqs[i];
const float vol_scale = scales[i];
const float mod_val = mod_values_ch[i];
// Note: + mod_val is specific to phase modulation
const double actual_phase = phase + mod_val;
const double pos = actual_phase * cur_size;
// Note: direct cast of negative doubles to uint32_t is undefined
const uint32_t pos1 = (uint32_t)(int32_t)floor(pos) & cur_size_mask;
const uint32_t pos2 = (pos1 + 1) & cur_size_mask;
const float item1 = cur_base[pos1];
const float item_diff = cur_base[pos2] - item1;
const double lerp_val = pos - floor(pos);
const double value =
(item1 + (lerp_val * item_diff)) * volume_factor * panning_factor;
out_buf_ch[i] += (float)value * vol_scale;
phase += freq * pitch_factor_inv_audio_rate;
// Normalise to range [0, 1)
if (phase >= 1)
{
phase -= 1;
// Don't bother updating the phase if our frequency is too high
if (phase >= 1)
phase = tone_state->phase[ch];
}
}
rassert(res_slice_start < res_slice_stop);
add_state->prev_mod[ch] = mod_values_ch[res_slice_stop - 1];
res_slice_start = res_slice_stop;
}
tone_state->phase[ch] = phase;
}
}
if (add->is_ramp_attack_enabled)
Proc_ramp_attack(vstate, 2, out_bufs, buf_start, buf_stop, dstate->audio_rate);
return buf_stop;
}
static void Compress_states_update(
Compress_state cstates[2],
const Proc_compress* compress,
Work_buffer* gain_wb,
Work_buffer* level_wbs[2],
const Work_buffer* in_wbs[2],
int32_t buf_start,
int32_t buf_stop,
int32_t audio_rate)
{
rassert(cstates != NULL);
rassert(compress != NULL);
rassert(gain_wb != NULL);
rassert(level_wbs != NULL);
rassert(level_wbs[0] != NULL);
rassert(level_wbs[1] != NULL);
rassert(in_wbs != NULL);
rassert(buf_start >= 0);
rassert(buf_stop > buf_start);
// Get levels
const float attack_mul =
(float)dB_to_scale(6 / (compress->attack * 0.001 * audio_rate));
const float release_mul =
(float)dB_to_scale(-6 / (compress->release * 0.001 * audio_rate));
for (int ch = 0; ch < 2; ++ch)
{
Compress_state* cstate = &cstates[ch];
rassert(cstate != NULL);
if (in_wbs[ch] == NULL)
continue;
float level = cstate->level;
float* levels = Work_buffer_get_contents_mut(level_wbs[ch]);
const float* in = Work_buffer_get_contents(in_wbs[ch]);
for (int32_t i = buf_start; i < buf_stop; ++i)
{
const float in_abs = fabsf(in[i]);
if (in_abs > level)
{
level *= attack_mul;
level = min(level, in_abs);
}
else
{
level *= release_mul;
level = max(level, max((float)MIN_LEVEL, in_abs));
}
levels[i] = level;
}
cstate->level = level;
}
if ((in_wbs[0] != NULL) && (in_wbs[1] != NULL))
{
// Get maximum levels
for (int32_t i = buf_start; i < buf_stop; ++i)
level_wbs[0] = max(level_wbs[0], level_wbs[1]);
}
const Work_buffer* applied_levels_wb =
(in_wbs[0] == NULL) ? level_wbs[1] : level_wbs[0];
const float* applied_levels = Work_buffer_get_contents(applied_levels_wb);
Work_buffer_clear(gain_wb, buf_start, buf_stop);
float* gains = Work_buffer_get_contents_mut(gain_wb);
for (int32_t i = buf_start; i < buf_stop; ++i)
gains[i] = 1.0f;
if (compress->upward_enabled)
{
// Apply upward compression
const double upward_threshold_dB =
compress->downward_enabled
? min(compress->upward_threshold, compress->downward_threshold)
: compress->upward_threshold;
const float threshold = (float)dB_to_scale(upward_threshold_dB);
const float inv_ratio = (float)(1.0 / compress->upward_ratio);
const float max_gain = (float)dB_to_scale(compress->upward_range);
for (int32_t i = buf_start; i < buf_stop; ++i)
{
const float level = applied_levels[i];
if (level < threshold)
{
const float diff = threshold / level;
const float gain = threshold / (powf(diff, inv_ratio) * level);
gains[i] = min(gain, max_gain);
}
}
}
if (compress->downward_enabled)
{
// Apply downward compression
const float threshold = (float)dB_to_scale(compress->downward_threshold);
const float inv_ratio = (float)(1.0 / compress->downward_ratio);
const float min_gain = (float)dB_to_scale(-compress->downward_range);
for (int32_t i = buf_start; i < buf_stop; ++i)
{
const float level = applied_levels[i];
if (level > threshold)
{
const float diff = level / threshold;
const float gain = (threshold * powf(diff, inv_ratio)) / level;
gains[i] = max(gain, min_gain);
}
}
}
return;
}
static void Freeverb_pstate_render_mixed(
Device_state* dstate,
const Work_buffers* wbs,
int32_t buf_start,
int32_t buf_stop,
double tempo)
{
assert(dstate != NULL);
assert(wbs != NULL);
assert(buf_start >= 0);
assert(tempo > 0);
Freeverb_pstate* fstate = (Freeverb_pstate*)dstate;
Proc_freeverb* freeverb = (Proc_freeverb*)dstate->device->dimpl;
// Get reflectivity parameter stream
float* refls = Device_state_get_audio_buffer_contents_mut(
dstate, DEVICE_PORT_TYPE_RECEIVE, PORT_IN_REFL);
if (refls == NULL)
{
refls = Work_buffers_get_buffer_contents_mut(wbs, FREEVERB_WB_FIXED_REFL);
const float fixed_refl = exp2(-5 / freeverb->reflect_setting);
for (int32_t i = buf_start; i < buf_stop; ++i)
refls[i] = fixed_refl;
}
else
{
// Convert reflectivity to the domain of our algorithm
static const float max_param_inv = -5.0 / 200.0;
static const float min_param_inv = -5.0 / 0.001;
for (int32_t i = buf_start; i < buf_stop; ++i)
{
const double orig_refl = refls[i];
const double param_inv = -5.0 / max(0, orig_refl);
const float refl = fast_exp2(clamp(param_inv, min_param_inv, max_param_inv));
refls[i] = refl;
}
}
// Get damp parameter stream
float* damps = Device_state_get_audio_buffer_contents_mut(
dstate, DEVICE_PORT_TYPE_RECEIVE, PORT_IN_DAMP);
if (damps == NULL)
{
damps = Work_buffers_get_buffer_contents_mut(wbs, FREEVERB_WB_FIXED_DAMP);
const float fixed_damp = freeverb->damp_setting * 0.01;
for (int32_t i = buf_start; i < buf_stop; ++i)
damps[i] = fixed_damp;
}
else
{
for (int32_t i = buf_start; i < buf_stop; ++i)
{
const float scaled_damp = damps[i] * 0.01f;
damps[i] = clamp(scaled_damp, 0, 1);
}
}
Work_buffer* in_wbs[] =
{
Device_state_get_audio_buffer(dstate, DEVICE_PORT_TYPE_RECEIVE, PORT_IN_AUDIO_L),
Device_state_get_audio_buffer(dstate, DEVICE_PORT_TYPE_RECEIVE, PORT_IN_AUDIO_R),
};
Work_buffer* out_wbs[] =
{
Device_state_get_audio_buffer(dstate, DEVICE_PORT_TYPE_SEND, PORT_OUT_AUDIO_L),
Device_state_get_audio_buffer(dstate, DEVICE_PORT_TYPE_SEND, PORT_OUT_AUDIO_R),
};
// TODO: figure out a cleaner way of dealing with the buffers
Work_buffer* workspace[] =
{
Work_buffers_get_buffer_mut(wbs, FREEVERB_WB_LEFT),
Work_buffers_get_buffer_mut(wbs, FREEVERB_WB_RIGHT),
};
// Get input data
if ((in_wbs[0] != NULL) && (in_wbs[1] != NULL))
{
Work_buffer_copy(workspace[0], in_wbs[0], buf_start, buf_stop);
Work_buffer_copy(workspace[1], in_wbs[1], buf_start, buf_stop);
}
else if ((in_wbs[0] == NULL) != (in_wbs[1] == NULL))
{
const Work_buffer* existing = (in_wbs[0] != NULL) ? in_wbs[0] : in_wbs[1];
Work_buffer_copy(workspace[0], existing, buf_start, buf_stop);
Work_buffer_copy(workspace[1], existing, buf_start, buf_stop);
}
else
{
Work_buffer_clear(workspace[0], buf_start, buf_stop);
Work_buffer_clear(workspace[1], buf_start, buf_stop);
}
float* ws[] =
{
Work_buffer_get_contents_mut(workspace[0]),
Work_buffer_get_contents_mut(workspace[1]),
};
// Apply reverb
{
float* comb_input =
Work_buffers_get_buffer_contents_mut(wbs, FREEVERB_WB_COMB_INPUT);
for (int32_t i = buf_start; i < buf_stop; ++i)
comb_input[i] = (ws[0][i] + ws[1][i]) * freeverb->gain;
for (int ch = 0; ch < 2; ++ch)
{
float* ws_buf = ws[ch];
for (int32_t i = buf_start; i < buf_stop; ++i)
ws_buf[i] = 0;
for (int comb = 0; comb < FREEVERB_COMBS; ++comb)
Freeverb_comb_process(
fstate->combs[ch][comb],
ws_buf,
comb_input,
refls,
damps,
buf_start,
buf_stop);
for (int allpass = 0; allpass < FREEVERB_ALLPASSES; ++allpass)
Freeverb_allpass_process(
fstate->allpasses[ch][allpass], ws_buf, buf_start, buf_stop);
}
for (int32_t i = buf_start; i < buf_stop; ++i)
{
ws[0][i] = ws[0][i] * freeverb->wet1 + ws[1][i] * freeverb->wet2;
ws[1][i] = ws[1][i] * freeverb->wet1 + ws[0][i] * freeverb->wet2;
}
}
// Copy results to outputs that exist
for (int ch = 0; ch < 2; ++ch)
{
if (out_wbs[ch] != NULL)
Work_buffer_copy(out_wbs[ch], workspace[ch], buf_start, buf_stop);
}
return;
}