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wav_player.c
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#include <stdio.h>
#include <math.h>
#include <string.h>
#include <sys/unistd.h>
#include <sys/stat.h>
#include "esp_err.h"
#include "esp_log.h"
#include "freertos/FreeRTOS.h"
#include "freertos/queue.h"
#include "freertos/task.h"
#include "driver/i2s.h"
#include "esp32-hal-log.h"
#include "esp32-hal-cpu.h"
#include "file_system.h"
#include "midi_in.h"
#include "file_system.h"
#include "ws_log.h"
#include "rpc.h"
#include "wav_player.h"
#include "fp.h"
static const char* TAG = "wav_player";
#define MAX_INT_16 32767
#define MIN_INT_16 -32768
#define wav_player_queue_SIZE 80
#define BLOCK_SIZE 512
#define BLOCKS_PER_READ 6
#define BYTES_PER_READ (BLOCKS_PER_READ * BLOCK_SIZE)
#define SAMPLES_PER_READ (BYTES_PER_READ / sizeof(int16_t))
#define DAC_BUFFER_SIZE_IN_SAMPLES 256
#define DAC_BUFFER_SIZE_IN_BYTES ( DAC_BUFFER_SIZE_IN_SAMPLES * sizeof(int16_t) )
#define LOOPS_PER_BUFFER ( BYTES_PER_READ / DAC_BUFFER_SIZE_IN_BYTES )
// #define NUM_BUFFERS 2 // debug
// #define NUM_BUFFERS 16 //
// #define NUM_BUFFERS 17 //
#define NUM_BUFFERS 18 // 1.0.x
// #define NUM_BUFFERS 19
// #define NUM_BUFFERS 20
// #define NUM_BUFFERS 30
#define DAMPEN_BITS 1
// #define MAX_READS_PER_LOOP 6
#define MAX_READS_PER_LOOP 5
// #define MAX_READS_PER_LOOP 4 // default
// #define MAX_READS_PER_LOOP 3
#define s15p16 int32_t
#define u16p16 uint32_t
// from midi.c
uint8_t channel_lut[16];
struct pan_t channel_pan[16];
uint8_t channel_vol[16];
uint8_t channel_exp[16];
uint8_t channel_attack[16];
uint8_t channel_release[16];
uint16_t channel_pitch_bend[16];
s15p16 channel_pitch_bend_factor[16];
extern struct metadata_t metadata;
bool mute = false;
struct asr_t {
size_t loop_start; // byte position to start reading into buf_head
size_t loop_end; // byte position to loop back to buf_head
size_t read_block; // block number to start reading from after buf_head
size_t offset; // num mono-samples to skip in the .read_block
size_t read_ptr; // num bytes read into buf_head
bool full; // the buffer has been filled previusly
};
struct asr_t make_asr_data(struct wav_lu_t wav)
{
struct asr_t asr;
asr.read_ptr = 0;
asr.loop_start = wav.loop_start * 2; // convert to mono samples
asr.loop_end = wav.loop_end * 2; // convert to mono samples
size_t end_buf_head = (asr.loop_start * 2) + (BYTES_PER_READ);
asr.read_block = end_buf_head / BLOCK_SIZE; // rounds down
size_t asr_offset_bytes = end_buf_head % BLOCK_SIZE; // this is the offset in bytes
asr.offset = asr_offset_bytes / sizeof(int16_t); // convert to mono-samples
asr.full = false;
return asr;
}
struct buf_t {
struct wav_lu_t wav_data;
struct wav_player_event_t wav_player_event;
struct wav_player_event_t next_wav_player_event;
struct asr_t asr;
struct vol_t stereo_volume;
struct vol_t target_stereo_volume;
int16_t *buffer_a;
int16_t *buffer_b;
int16_t *buffer_head;
size_t read_block;
size_t wav_position;
size_t size;
u16p16 sample_pointer;
size_t fade_counter;
uint8_t volume;
uint8_t voice;
int8_t fade;
uint8_t current_buf;
uint8_t pause_state :2;
uint8_t free :1;
uint8_t done :1;
uint8_t full :1;
uint8_t pruned :1;
};
esp_err_t ret;
QueueHandle_t wav_player_queue;
QueueHandle_t dac_queue_handle;
i2s_event_t dac_event;
// declare local function prototypes
void wav_player_pause(void);
// declare function prototypes from emmc.c
esp_err_t emmc_read(void *dst, size_t start_sector, size_t sector_count);
// declare function prototypes from dac.c
esp_err_t dac_write(const void *src, size_t size, size_t *bytes_written);
esp_err_t dac_write_int(const void *src, size_t size, size_t *bytes_written);
void dac_pause(void);
void dac_resume(void);
struct buf_t bufs[NUM_BUFFERS];
int *output_buf;
int16_t *output_buf_16;
struct wav_player_event_t wav_player_event;
int new_midi = 0;
int new_midi_buf = -1;
struct response_curve_t {
s15p16 val;
uint8_t fade_val;
};
struct response_curve_t lin_response_lut[128];
struct response_curve_t sqrt_response_lut[128];
struct response_curve_t inv_sqrt_response_lut[128];
int16_t sample;
void init_buffs(void)
{
output_buf = (int *)ps_malloc( DAC_BUFFER_SIZE_IN_SAMPLES * sizeof(int) );
if(output_buf==NULL)
{
log_e("failed to alloc output buf");
}
output_buf_16 = (int16_t *)malloc( DAC_BUFFER_SIZE_IN_SAMPLES * sizeof(int16_t) );
if(output_buf_16==NULL)
{
log_e("failed to alloc output buf 16");
}
for(uint8_t i=0; i<NUM_BUFFERS; i++)
{
// flags
bufs[i].free = 1;
bufs[i].done = 0;
bufs[i].full = 0;
bufs[i].pruned = 0;
bufs[i].pause_state = PAUSE_NONE;
bufs[i].current_buf = 0;
// uint8_t's
bufs[i].volume=127;
bufs[i].fade=0;
// size_t's
bufs[i].read_block = 0;
bufs[i].wav_position = 0;
bufs[i].size = 0;
bufs[i].sample_pointer = 0;
bufs[i].fade_counter = 0;
// buffers
bufs[i].buffer_a = (int16_t *)malloc(BYTES_PER_READ);
bufs[i].buffer_b = (int16_t *)malloc(BYTES_PER_READ);
bufs[i].buffer_head = (int16_t *)ps_malloc(BYTES_PER_READ);
if(bufs[i].buffer_a==NULL || bufs[i].buffer_b == NULL || bufs[i].buffer_head == NULL)
{
log_e("failed to alloc buffers at %d",i);
}
}
for(int i=0;i<DAC_BUFFER_SIZE_IN_SAMPLES;i++)
{
output_buf[i]=0;
output_buf_16[i]=0;
}
log_i("%d buffers were initialized", NUM_BUFFERS);
}
void free_bufs(void)
{
for(int i=0;i<NUM_BUFFERS;i++)
{
free(bufs[i].buffer_a);
free(bufs[i].buffer_b);
free(bufs[i].buffer_head);
}
}
void init_response_lut_fade_pair(struct response_curve_t *dest, struct response_curve_t *pair)
{
for(int i=0;i<128;i++)
{
// find the nearest but smaller pair, so that the linear fade out starts at the best spot
for(int j=0;j<128;j++)
{
if(pair[j+1].val > dest[i].val)
{
dest[i].fade_val = j;
break;
}
// force the last one
if(j == 127)
{
dest[i].fade_val = j;
}
}
}
}
void update_pitch_bends(void)
{
for(int i=0; i<16; i++)
{
uint16_t pitch_bend = channel_pitch_bend[i]; // 14 bit
s15p16 bend = (pitch_bend << 16) / 8192.0 - ( 1 << 16);
// s15p16 semitones = bend >= 0 ? 12 * bend : 12 * bend;
s15p16 semitones = bend >= 0 ? metadata.pitch_bend_semitones_up * bend : metadata.pitch_bend_semitones_down * bend;
s15p16 pitch_factor = fxexp2_s15p16(semitones / 12);
channel_pitch_bend_factor[i] = pitch_factor;
}
}
void convert_buf_linear(int buf)
{
switch(bufs[buf].wav_data.response_curve){
case RESPONSE_SQUARE_ROOT:
bufs[buf].stereo_volume.left = sqrt_response_lut[bufs[buf].stereo_volume.left].fade_val;
bufs[buf].stereo_volume.right = sqrt_response_lut[bufs[buf].stereo_volume.right].fade_val;
break;
case RESPONSE_INV_SQUARE_ROOT:
bufs[buf].stereo_volume.left = inv_sqrt_response_lut[bufs[buf].stereo_volume.left].fade_val;
bufs[buf].stereo_volume.right = inv_sqrt_response_lut[bufs[buf].stereo_volume.right].fade_val;
break;
default:
break;
}
bufs[buf].wav_data.response_curve = RESPONSE_LINEAR;
}
void init_response_luts(void)
{
for(int i=0;i<128;i++)
{
lin_response_lut[i].val = (i / 127.0) * 0x10000;
sqrt_response_lut[i].val = (sqrt(i * 127.0) / 127) * 0x10000;
inv_sqrt_response_lut[i].val = (pow(i / 127.0, 2)) * 0x10000;
}
init_response_lut_fade_pair(sqrt_response_lut, lin_response_lut);
init_response_lut_fade_pair(inv_sqrt_response_lut, lin_response_lut);
init_response_lut_fade_pair(lin_response_lut, lin_response_lut);
}
int16_t IRAM_ATTR scale_sample (int16_t in, uint8_t vol)
{
// return (int16_t)(in * lin_response_lut[vol].val);
return (int16_t)((in * lin_response_lut[vol].val) >> 16);
}
int16_t IRAM_ATTR scale_sample_sqrt (int16_t in, uint8_t vol)
{
// return (int16_t)(in * sqrt_response_lut[vol].val);
return (int16_t)((in * sqrt_response_lut[vol].val) >> 16);
}
int16_t IRAM_ATTR scale_sample_inv_sqrt (int16_t in, uint8_t vol)
{
// return (int16_t)(in * inv_sqrt_response_lut[vol].val);
return (int16_t)((in * inv_sqrt_response_lut[vol].val) >> 16);
}
int16_t IRAM_ATTR scale_sample_clamped_16(int in, uint8_t volume)
{
int16_t out = (in > MAX_INT_16) ? MAX_INT_16 : (in < MIN_INT_16) ? MIN_INT_16 : in;
// return (int16_t)(out * lin_response_lut[volume].val);
return (int16_t)((out * lin_response_lut[volume].val) >> 16);
}
bool is_playing(uint8_t voice, uint8_t note)
{
for(int i=0;i<NUM_BUFFERS;i++)
{
if((bufs[i].free == false) && (bufs[i].wav_player_event.voice == voice) && (bufs[i].wav_player_event.note == note))
{
return true;
}
}
return false;
}
void stop_wav(uint8_t voice, uint8_t note)
{
struct wav_player_event_t wav_player_event;
wav_player_event.code = MIDI_NOTE_OFF;
wav_player_event.voice = voice;
wav_player_event.velocity = 0;
wav_player_event.note = note;
xQueueSendToBack(wav_player_queue, &wav_player_event, portMAX_DELAY);
}
void play_wav(uint8_t voice, uint8_t note, uint8_t velocity)
{
struct wav_player_event_t wav_player_event;
wav_player_event.code = MIDI_NOTE_ON;
wav_player_event.voice = voice;
wav_player_event.velocity = velocity;
wav_player_event.note = note;
wav_player_event.channel = 0;
xQueueSendToBack(wav_player_queue, &wav_player_event, portMAX_DELAY);
}
void toggle_wav(uint8_t voice, uint8_t note, uint8_t velocity)
{
if(is_playing(voice, note))
{
stop_wav(voice, note);
}
else
{
play_wav(voice, note, velocity);
}
}
int prune(uint8_t priority)
{
int candidate = -1;
for(int i=0;i<NUM_BUFFERS;i++)
{
if(bufs[i].pruned || bufs[i].wav_data.play_back_mode == ASR_LOOP)
{
continue;
}
if(candidate == -1 && (bufs[i].wav_data.priority <= priority))
{
// this is the first candidate
candidate = i;
}
else
{
// log_i("i: %d, c: %d",bufs[i].wav_data.priority,bufs[candidate].wav_data.priority);
if(bufs[i].wav_data.priority < bufs[candidate].wav_data.priority)
{
candidate = i;
// log_i("didnt check wav position");
}
else if(
(bufs[i].wav_data.priority == bufs[candidate].wav_data.priority) &&
(bufs[i].wav_position > bufs[candidate].wav_position)
)
{
// log_i("checked wav position");
candidate = i;
}
}
}
// log_i("pruned buf %d",candidate);
return candidate;
}
void IRAM_ATTR update_stereo_volume(uint8_t buf, enum stereo_mode stereo_mode)
{
uint8_t chan = bufs[buf].wav_player_event.channel;
uint32_t left = channel_vol[chan] * channel_exp[chan] * channel_pan[chan].left_vol * bufs[buf].volume;
uint32_t right = channel_vol[chan] * channel_exp[chan] * channel_pan[chan].right_vol * bufs[buf].volume;
// bufs[buf].stereo_volume.left = (uint8_t)(left / 2048383); // 127*127*127
// bufs[buf].stereo_volume.right = (uint8_t)(right / 2048383);
bufs[buf].stereo_volume.left = stereo_mode == STEREO_MODE_STEREO ? ((uint8_t)(left / 2048383)) : stereo_mode == STEREO_MODE_MONO_LEFT ? 127 : 0; // 127*127*127
bufs[buf].stereo_volume.right = stereo_mode == STEREO_MODE_STEREO ? ((uint8_t)(right / 2048383)) : stereo_mode == STEREO_MODE_MONO_LEFT ? 0 : 127;
// copy to target
bufs[buf].target_stereo_volume.left = bufs[buf].stereo_volume.left;
bufs[buf].target_stereo_volume.right = bufs[buf].stereo_volume.right;
}
int16_t IRAM_ATTR interpolate(int16_t a, int16_t b, s15p16 frac)
{
int32_t y = b - a; // delta y,
int32_t delta = (frac * y) >> 16; // execute fixed point multiply to do the linear interpolation
return(a + delta); // add the base
}
void IRAM_ATTR wav_player_task(void* pvParameters)
{
static size_t bytes_to_dma = 0;
size_t base_index;
int16_t *buf_pointer;
int num_reads = 0;
bool abort_note = false;
if((wav_player_queue = xQueueCreate(wav_player_queue_SIZE, sizeof(struct wav_player_event_t)))==pdFAIL)
{
log_e("failed to creat wav_player_queue");
}
log_i("wav player running on core %u",xPortGetCoreID());
for(;;)
{
// check for midi events and add try to place them into the buffers
if(xQueueReceive(wav_player_queue, &wav_player_event, 0))
{
abort_note = false;
// fetch the file
struct wav_lu_t new_wav = get_file_t_from_lookup_table(wav_player_event.voice, wav_player_event.note, wav_player_event.velocity);
// check that there is a wav file there
if(new_wav.empty == 1)
{
abort_note = 1;
// log_i("found empty");
// check for PAUSE RESETS
if(
(wav_player_event.code == MIDI_NOTE_ON) &&
(new_wav.mute_group > 0)
){
for(int8_t i=0; i<NUM_BUFFERS; i++)
{
if(
(bufs[i].free == 0) &&
(bufs[i].wav_data.mute_group == new_wav.mute_group) &&
(
(bufs[i].wav_data.play_back_mode == PAUSE) ||
(bufs[i].wav_data.play_back_mode == PAUSE_LOOP) ||
(bufs[i].wav_data.play_back_mode == PAUSE_ASR)
)
)
{
// log_i("pruned because new wav");
switch(bufs[i].pause_state){
case PAUSE_NONE:
case PAUSE_START:
case PAUSE_RESUMING:
bufs[i].fade = FADE_OUT;
bufs[i].pruned = 1;
break;
case PAUSE_PAUSED:
bufs[i].free = 1;
bufs[i].done = 0;
bufs[i].current_buf = 0;
bufs[i].wav_position = 0;
bufs[i].sample_pointer = 0;
bufs[i].fade_counter = 0;
bufs[i].fade = FADE_NORMAL;
bufs[i].pause_state = PAUSE_NONE;
break;
default:break;
}
}
}
}
};
// secret midi note to trigger pause
if(wav_player_event.voice == 0xFF) wav_player_pause();
// check if it is a disguised note-off
if(wav_player_event.code == MIDI_NOTE_ON && wav_player_event.velocity == 0) wav_player_event.code = MIDI_NOTE_OFF;
// check if this is a retrigger
if(wav_player_event.code == MIDI_NOTE_ON && !abort_note && wav_player_event.velocity != 0)
{
for(int b=0;b<NUM_BUFFERS;b++)
{
if(
bufs[b].wav_player_event.voice == wav_player_event.voice &&
bufs[b].wav_player_event.note == wav_player_event.note &&
bufs[b].free == 0 &&
bufs[b].fade != FADE_OUT
)
{
if(bufs[b].pause_state == PAUSE_PAUSED)
{
//resume
// log_i("resume");
bufs[b].pause_state = PAUSE_RESUMING;
bufs[b].fade = FADE_IN_INIT;
abort_note = 1;
}
else
{
// it is a retrigger, do the right thing
switch(bufs[b].wav_data.retrigger_mode){
case RESTART:
bufs[b].fade = FADE_OUT;
bufs[b].pruned = 1;
// log_i("pruned because RESTART");
break;
case NOTE_OFF:
bufs[b].fade = FADE_OUT;
abort_note = 1;
if(
bufs[b].wav_data.play_back_mode == PAUSE ||
bufs[b].wav_data.play_back_mode == PAUSE_LOOP ||
bufs[b].wav_data.play_back_mode == PAUSE_ASR
)
{
// log_i("pausing");
bufs[b].pause_state = PAUSE_START;
}
break;
case NONE:
abort_note = 1;
break;
default:
break;
}
}
break;
}
}
}
// check mute groups
if(wav_player_event.code == MIDI_NOTE_ON && !abort_note && new_wav.mute_group > 0)
{
for(int8_t i=0; i<NUM_BUFFERS; i++)
{
if(
(bufs[i].free == 0) &&
(bufs[i].wav_data.mute_group == new_wav.mute_group)
)
{
bufs[i].fade = FADE_OUT;
bufs[i].pruned = 1;
// log_i("pruned because mute group");
}
}
}
// try to play the note
if(wav_player_event.code == MIDI_NOTE_ON && !abort_note)
{
// find a free buffer
for(int8_t i=0; i<=NUM_BUFFERS; i++)
{
if(i == NUM_BUFFERS) // none are free
{
// prune
int to_prune = prune(new_wav.priority);
if(to_prune == -1)
{
log_v("note blocked because bufs full of higher priorities");
break;
}
else
{
bufs[to_prune].next_wav_player_event = wav_player_event;
bufs[to_prune].fade = FADE_OUT;
bufs[to_prune].pruned = 1;
// log_i("pruned because out of buffers");
break;
}
}
if(bufs[i].free == 1)
{
// rpc_out(RPC_NOTE_ON, bufs[i].voice, wav_player_event.note, wav_player_event.velocity);
new_midi = 1;
new_midi_buf = i;
// wlog_d("adding to buffer %d",i);
bufs[i].wav_data = new_wav;
if(bufs[i].wav_data.length == 0){
// its a null_wav_file because it was a rack and it wasn't within the velocity ranges
break;
}
bufs[i].wav_player_event = wav_player_event;
bufs[i].voice = wav_player_event.voice;
bufs[i].free = 0;
bufs[i].done = 0;
bufs[i].fade = FADE_NORMAL;
bufs[i].full = 0;
bufs[i].current_buf = 0;
bufs[i].sample_pointer = 0;
bufs[i].fade_counter = 0;
bufs[i].read_block = bufs[i].wav_data.start_block;
bufs[i].wav_position = 0;
bufs[i].size = bufs[i].wav_data.length / sizeof(int16_t);
if(
((new_wav.isRack == -2) && (new_wav.velocity_mode == VELOCITY_MODE_FIXED)) ||
new_wav.response_curve == RESPONSE_FIXED
) // it's a rack member with fixed velocity or a fixed volume file
{
bufs[i].volume = 127;
}
else // its not a rack member or a fixed volume file
{
bufs[i].volume = wav_player_event.velocity;
}
if(
new_wav.play_back_mode == ASR_LOOP ||
new_wav.play_back_mode == PAUSE_ASR
) // calculate the ASR data if needed
{
bufs[i].asr = make_asr_data(new_wav);
}
else // setup for attack if needed
{
if(channel_attack[bufs[i].wav_player_event.channel] > 0) // use attack
{
bufs[i].fade = FADE_IN_INIT; // 0 means start attack
}
}
break;
}
}
}
else if (wav_player_event.code == MIDI_NOTE_OFF){
for(int b=0;b<NUM_BUFFERS;b++)
{
if(
bufs[b].wav_player_event.voice == wav_player_event.voice &&
bufs[b].wav_player_event.note == wav_player_event.note &&
bufs[b].free == 0 &&
(
// bufs[b].wav_data.note_off_meaning == HALT
bufs[b].wav_data.note_off_meaning == HALT ||
bufs[b].wav_data.note_off_meaning == RELEASE
)
)
{
bufs[b].fade = FADE_OUT;
if(
bufs[b].wav_data.play_back_mode == PAUSE ||
bufs[b].wav_data.play_back_mode == PAUSE_LOOP ||
bufs[b].wav_data.play_back_mode == PAUSE_ASR
)
{
// log_i("pausing");
bufs[b].pause_state = PAUSE_START;
}
// bufs[b].done=1;
// break; // comment out to stop them all, leave it uncommented to break just the first one that matches
}
}
}
else if(wav_player_event.code == MIDI_CC)
{
if(wav_player_event.note == MIDI_CC_MUTE)
{
for(int buf=0; buf<NUM_BUFFERS; buf++)
{
if(bufs[buf].wav_player_event.channel == wav_player_event.channel)
{
bufs[buf].done = true;
}
}
}
}
}
num_reads = 0;
if(new_midi == 1) // if there is a new midi message, read into that buffer for sure
{
// if its a looped sample, read into the buffer_head
if(bufs[new_midi_buf].wav_data.play_back_mode == LOOP)
{
ESP_ERROR_CHECK(emmc_read(bufs[new_midi_buf].buffer_head, bufs[new_midi_buf].read_block ,BLOCKS_PER_READ ));
bufs[new_midi_buf].current_buf = 2;
}
else
{
ESP_ERROR_CHECK(emmc_read(bufs[new_midi_buf].buffer_a, bufs[new_midi_buf].read_block ,BLOCKS_PER_READ ));
}
num_reads++;
bufs[new_midi_buf].read_block += BLOCKS_PER_READ;
new_midi = 0;
}
// read into all the other buffers that need it, up to MAX_READS_PER_LOOP
for(int i=0;i<NUM_BUFFERS;i++)
{
if(
bufs[i].free == 0 &&
bufs[i].full == 0 &&
num_reads < MAX_READS_PER_LOOP &&
i != new_midi_buf
)
{
buf_pointer = bufs[i].current_buf == 0 ? bufs[i].buffer_b : bufs[i].buffer_a;
ESP_ERROR_CHECK(emmc_read(buf_pointer, bufs[i].read_block , BLOCKS_PER_READ ));
num_reads++;
bufs[i].read_block += BLOCKS_PER_READ;
bufs[i].full = 1; // now the buffer is full
}
}
new_midi_buf=-1;
// update all the pitch bend states only once per render cycle, it is expensive!
update_pitch_bends();
// sum the next section of each buffer and send to DAC buffer
for(int buf = 0; buf < NUM_BUFFERS; buf++)
{
if(
(bufs[buf].free == 0) &&
(bufs[buf].pause_state != PAUSE_PAUSED)
)
{
buf_pointer = bufs[buf].current_buf == 0 ? bufs[buf].buffer_a : bufs[buf].current_buf == 1 ? bufs[buf].buffer_b : bufs[buf].buffer_head;
switch(bufs[buf].wav_data.play_back_mode){
case ONE_SHOT:
case PAUSE:
{
size_t remaining = bufs[buf].size - bufs[buf].wav_position;
u16p16 step = channel_pitch_bend_factor[bufs[buf].wav_player_event.channel];
if(bufs[buf].fade == FADE_NORMAL) // dont update stereo volume while fading
{
update_stereo_volume(buf, bufs[buf].wav_data.stereo_mode);
}
else if(bufs[buf].pruned && bufs[buf].wav_data.response_curve != RESPONSE_LINEAR) // starting a fade but the buf is not a linear response
{
convert_buf_linear(buf);
}
int fade_factor = (bufs[buf].pruned || (bufs[buf].pause_state == PAUSE_RESUMING)) ? 4 // fast fadeout/fadein
: bufs[buf].fade == FADE_OUT ? 4 + (channel_release[bufs[buf].wav_player_event.channel] * FADE_FACTOR_MULTIPLIER) // release
: 4 + (channel_attack[bufs[buf].wav_player_event.channel] * FADE_FACTOR_MULTIPLIER); // attack
for(int i=0; i<DAC_BUFFER_SIZE_IN_SAMPLES; i += 2)
{
uint32_t idx = (bufs[buf].sample_pointer >> 16) * 2; // just the whole part
s15p16 frac = bufs[buf].sample_pointer & 0x0000FFFF; // just the fractional part
bool will_ovrflw_buf = idx > ( SAMPLES_PER_READ - 4 );
int16_t *ovflw_buf_pointer = bufs[buf].current_buf == 0 ? bufs[buf].buffer_b : bufs[buf].buffer_a;
int16_t sample_left_a = buf_pointer[idx];
int16_t sample_right_a = buf_pointer[idx+1];
int16_t sample_left_b = will_ovrflw_buf ? ovflw_buf_pointer[0] : buf_pointer[idx+2];
int16_t sample_right_b = will_ovrflw_buf ? ovflw_buf_pointer[1] : buf_pointer[idx+3];
int16_t sample_left = interpolate(sample_left_a, sample_left_b, frac);
int16_t sample_right = interpolate(sample_right_a, sample_right_b, frac);
sample_left =
bufs[buf].wav_data.response_curve == RESPONSE_LINEAR ?
scale_sample(sample_left, bufs[buf].stereo_volume.left) :
bufs[buf].wav_data.response_curve == RESPONSE_SQUARE_ROOT ?
scale_sample_sqrt(sample_left, bufs[buf].stereo_volume.left) :
scale_sample_inv_sqrt(sample_left, bufs[buf].stereo_volume.left);
sample_right =
bufs[buf].wav_data.response_curve == RESPONSE_LINEAR ?
scale_sample(sample_right, bufs[buf].stereo_volume.right) :
bufs[buf].wav_data.response_curve == RESPONSE_SQUARE_ROOT ?
scale_sample_sqrt(sample_right, bufs[buf].stereo_volume.right) :
scale_sample_inv_sqrt(sample_right, bufs[buf].stereo_volume.right);
output_buf[i] += (sample_left >> DAMPEN_BITS);
output_buf[i + 1] += (sample_right >> DAMPEN_BITS);
bufs[buf].sample_pointer += step;
size_t written = (bufs[buf].sample_pointer >> 16) * 2;
if(written >= remaining)
{
bufs[buf].done = true;
break;
}
if(written >= SAMPLES_PER_READ)
{
buf_pointer = bufs[buf].current_buf == 0 ? bufs[buf].buffer_b : bufs[buf].buffer_a;
bufs[buf].current_buf = bufs[buf].current_buf == 0 ? 1 : 0;
bufs[buf].sample_pointer -= ( SAMPLES_PER_READ * 0x8000);
bufs[buf].full = 0;
bufs[buf].wav_position += SAMPLES_PER_READ;
}
if((bufs[buf].fade != FADE_NORMAL) && ((bufs[buf].fade_counter % fade_factor < 2)))
{
if( (bufs[buf].fade == FADE_OUT))
{
bufs[buf].stereo_volume.right -= (bufs[buf].stereo_volume.right > 0); // decriment unless 0
bufs[buf].stereo_volume.left -= (bufs[buf].stereo_volume.left > 0);
if((bufs[buf].stereo_volume.right == 0) && (bufs[buf].stereo_volume.left == 0)) // fade complete
{
bufs[buf].done = true;
break;
}
}
else
{
bufs[buf].fade += (bufs[buf].fade < 127);
bufs[buf].stereo_volume.right = bufs[buf].fade < bufs[buf].target_stereo_volume.right ? bufs[buf].fade : bufs[buf].target_stereo_volume.right;
bufs[buf].stereo_volume.left = bufs[buf].fade < bufs[buf].target_stereo_volume.left ? bufs[buf].fade : bufs[buf].target_stereo_volume.left;
if(
(bufs[buf].stereo_volume.right == bufs[buf].target_stereo_volume.right) &&
(bufs[buf].stereo_volume.left == bufs[buf].target_stereo_volume.left)) // attack done
{
bufs[buf].fade = FADE_NORMAL;
bufs[buf].fade_counter = 0;
bufs[buf].pause_state = PAUSE_NONE;
// log_i("pause none");
}
}
}
bufs[buf].fade_counter += 2;
}
break;
}
case LOOP :
case PAUSE_LOOP :
{
size_t remaining = bufs[buf].size - bufs[buf].wav_position;
u16p16 step = channel_pitch_bend_factor[bufs[buf].wav_player_event.channel];
if(bufs[buf].done) // if it fades out, stop asap
break;
if(bufs[buf].fade == FADE_NORMAL) // only update the volume when NOT fading-out or about to start fading-in
{
update_stereo_volume(buf, bufs[buf].wav_data.stereo_mode);
}
else if(bufs[buf].pruned && (bufs[buf].wav_data.response_curve != RESPONSE_LINEAR)) // starting a fade but the buf is not a linear response
{
convert_buf_linear(buf);
}
int fade_factor = (bufs[buf].pruned || (bufs[buf].pause_state == PAUSE_RESUMING)) ? 4 // fast fadeout/fadein
: bufs[buf].fade == FADE_OUT ? 4 + (channel_release[bufs[buf].wav_player_event.channel] * FADE_FACTOR_MULTIPLIER) // release
: 4 + (channel_attack[bufs[buf].wav_player_event.channel] * FADE_FACTOR_MULTIPLIER); // attack
for(int i=0; i<DAC_BUFFER_SIZE_IN_SAMPLES; i += 2)
{
uint32_t idx = (bufs[buf].sample_pointer >> 16) * 2; // just the whole part, *2 because stereo
s15p16 frac = bufs[buf].sample_pointer & 0x0000FFFF; // just the fractional part
bool will_ovrflw_loop = idx > ( remaining - 4 );
bool will_ovrflw_buf = idx > ( SAMPLES_PER_READ - 4 );
int16_t *ovflw_buf_pointer = will_ovrflw_loop ? bufs[buf].buffer_head : bufs[buf].current_buf == 0 ? bufs[buf].buffer_b : bufs[buf].buffer_a;
int16_t sample_left_a = buf_pointer[idx];
int16_t sample_right_a = buf_pointer[idx+1];
int16_t sample_left_b = (will_ovrflw_loop || will_ovrflw_buf) ? ovflw_buf_pointer[0] : buf_pointer[idx+2];
int16_t sample_right_b = (will_ovrflw_loop || will_ovrflw_buf) ? ovflw_buf_pointer[1] : buf_pointer[idx+3];
int16_t sample_left = interpolate(sample_left_a, sample_left_b, frac);
int16_t sample_right = interpolate(sample_right_a, sample_right_b, frac);
sample_left = bufs[buf].wav_data.response_curve == RESPONSE_LINEAR ?
scale_sample(sample_left, bufs[buf].stereo_volume.left) :
bufs[buf].wav_data.response_curve == RESPONSE_SQUARE_ROOT ?
scale_sample_sqrt(sample_left, bufs[buf].stereo_volume.left) :
scale_sample_inv_sqrt(sample_left, bufs[buf].stereo_volume.left);
sample_right = bufs[buf].wav_data.response_curve == RESPONSE_LINEAR ?
scale_sample(sample_right, bufs[buf].stereo_volume.right) :
bufs[buf].wav_data.response_curve == RESPONSE_SQUARE_ROOT ?
scale_sample_sqrt(sample_right, bufs[buf].stereo_volume.right) :
scale_sample_inv_sqrt(sample_right, bufs[buf].stereo_volume.right);
output_buf[i] += (sample_left >> DAMPEN_BITS);
output_buf[i + 1] += (sample_right >> DAMPEN_BITS);
bufs[buf].sample_pointer += step;
size_t written = (bufs[buf].sample_pointer >> 16) * 2;
if(written >= remaining) // the wav needs to loop now
{
buf_pointer = bufs[buf].buffer_head;
bufs[buf].current_buf = 2;
bufs[buf].wav_position = 0;
// bufs[buf].sample_pointer &= 0x0000FFFF; // keep the fractional part
bufs[buf].sample_pointer -= (remaining * 0x8000); // keep the fractional part
bufs[buf].read_block = bufs[buf].wav_data.start_block + BLOCKS_PER_READ; // next read can skip buffer_head
bufs[buf].full = 0;
remaining = bufs[buf].size;
}
else if(written >= SAMPLES_PER_READ) // out of buffer but more to the wav
{
buf_pointer = bufs[buf].current_buf == 0 ? bufs[buf].buffer_b : bufs[buf].buffer_a;
bufs[buf].current_buf = bufs[buf].current_buf == 0 ? 1 : 0;
bufs[buf].sample_pointer -= ( SAMPLES_PER_READ * 0x8000);
bufs[buf].full = 0;
bufs[buf].wav_position += SAMPLES_PER_READ;
}
if((bufs[buf].fade != FADE_NORMAL) && (bufs[buf].fade_counter % fade_factor < 2))
{
if( bufs[buf].fade == FADE_OUT )
{
bufs[buf].stereo_volume.right -= (bufs[buf].stereo_volume.right > 0); // decriment unless 0
bufs[buf].stereo_volume.left -= (bufs[buf].stereo_volume.left > 0);
if((bufs[buf].stereo_volume.right == 0) && (bufs[buf].stereo_volume.left == 0)) // fade complete
{
bufs[buf].done = true;
break;
}
}
else
{
bufs[buf].fade += (bufs[buf].fade < 127);
bufs[buf].stereo_volume.right = bufs[buf].fade < bufs[buf].target_stereo_volume.right ? bufs[buf].fade : bufs[buf].target_stereo_volume.right;
bufs[buf].stereo_volume.left = bufs[buf].fade < bufs[buf].target_stereo_volume.left ? bufs[buf].fade : bufs[buf].target_stereo_volume.left;
if(
(bufs[buf].stereo_volume.right == bufs[buf].target_stereo_volume.right) &&
(bufs[buf].stereo_volume.left == bufs[buf].target_stereo_volume.left)) // attack done
{
bufs[buf].fade = FADE_NORMAL;
bufs[buf].fade_counter = 0;
}
}
}
bufs[buf].fade_counter += 2;
}
break;
}
/*
ANATOMY OF ASR LOOP
|<--attack-->|<----------------------sustain--------------------->|<---release--->|
|<------buffer_head------>|
| | | | | | | | | | | | | | | <-natural buffer alignment
^
^ | ^
| asr.read_block |
asr.loop_start asr.loop_end
|<->|
^
|
asr.offset
*/
case ASR_LOOP :
case PAUSE_ASR :
{
u16p16 step = channel_pitch_bend_factor[bufs[buf].wav_player_event.channel];
if(bufs[buf].fade == FADE_NORMAL) // only update the volume when NOT fading-out or about to start fading-in
{
update_stereo_volume(buf, bufs[buf].wav_data.stereo_mode);
}
else if(bufs[buf].pruned && (bufs[buf].wav_data.response_curve != RESPONSE_LINEAR)) // starting a fade but the buf is not a linear response
{
convert_buf_linear(buf);
}
int fade_factor = bufs[buf].pruned ? 4 : 4 + channel_release[bufs[buf].wav_player_event.channel];
int i = 0;
while(i < DAC_BUFFER_SIZE_IN_SAMPLES)
{
uint32_t idx = (bufs[buf].sample_pointer >> 16) * 2;
s15p16 frac = bufs[buf].sample_pointer & 0x0000FFFF;
uint32_t position = bufs[buf].wav_position + idx;
int section;
uint32_t remaining;
bool should_copy = false;
if( position < bufs[buf].asr.loop_start )
{
section = ASR_ATTACK;
remaining = bufs[buf].asr.loop_start - bufs[buf].wav_position;
}
else if(position < (bufs[buf].asr.loop_start + SAMPLES_PER_READ))
{
section = ASR_HEAD;
remaining = (bufs[buf].asr.loop_start + SAMPLES_PER_READ) - bufs[buf].wav_position;
should_copy = bufs[buf].asr.full == false;
}
else if(position < bufs[buf].asr.loop_end)
{
section = ASR_SUSTAIN;
remaining = bufs[buf].asr.loop_end - bufs[buf].wav_position;
}
else
{
section = ASR_RELEASE;
remaining = bufs[buf].size - bufs[buf].wav_position;
}
while(idx < remaining)
{