int64_t ceil_p2(int64_t x)
{
assert(x > 0);
if (is_p2(x))
return x;
return next_p2(x);
}
void Device_port_groups_init(Device_port_groups groups, int first_group_size, ...)
{
rassert(groups != NULL);
rassert(first_group_size >= 0);
rassert(implies(first_group_size > 0, is_p2(first_group_size)));
rassert(first_group_size <= WORK_BUFFER_SUB_COUNT_MAX);
for (int i = 0; i < PORT_GROUPS_MAX; ++i)
groups[i] = 0;
va_list args;
va_start(args, first_group_size);
groups[0] = (int8_t)first_group_size;
int size_count = 0;
if (first_group_size > 0)
{
for (size_count = 1; size_count < PORT_GROUPS_MAX; ++size_count)
{
const int size = va_arg(args, int);
rassert(size >= 0);
rassert(size <= WORK_BUFFER_SUB_COUNT_MAX);
rassert(implies(size > 0, is_p2(size)));
groups[size_count] = (int8_t)size;
if (groups[size_count] == 0)
break;
}
}
if (size_count == PORT_GROUPS_MAX)
{
const int zero = va_arg(args, int);
rassert(zero == 0);
}
va_end(args);
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;
}
