#include "shared-bindings/audiomp3/MP3Decoder.h"

#include <stdint.h>

#include <string.h>

#include <math.h>

#include "py/mperrno.h"

#include "py/runtime.h"

#include "shared-module/audiomp3/MP3Decoder.h"

#include "supervisor/shared/translate.h"

#include "supervisor/background_callback.h"

#include "lib/mp3/src/mp3common.h"

#define MAX_BUFFER_LEN (MAX_NSAMP * MAX_NGRAN * MAX_NCHAN * sizeof(int16_t))

STATIC bool mp3file_update_inbuf_always(audiomp3_mp3file_obj_t *self) {

// If we didn't previously reach the end of file, we can try reading now

if (!self->eof) {

// Move the unconsumed portion of the buffer to the start

uint8_t *end_of_buffer = self->inbuf + self->inbuf_length;

uint8_t *new_end_of_data = self->inbuf + self->inbuf_length - self->inbuf_offset;

memmove(self->inbuf, self->inbuf + self->inbuf_offset,

self->inbuf_length - self->inbuf_offset);

self->inbuf_offset = 0;

UINT to_read = end_of_buffer - new_end_of_data;

UINT bytes_read = 0;

memset(new_end_of_data, 0, to_read);

if (f_read(&self->file->fp, new_end_of_data, to_read, &bytes_read) != FR_OK) {

self->eof = true;

mp_raise_OSError(MP_EIO);

}

if (bytes_read == 0) {

self->eof = true;

}

if (to_read != bytes_read) {

new_end_of_data += bytes_read;

memset(new_end_of_data, 0, end_of_buffer - new_end_of_data);

}

}

// Return true iff there are at least some useful bytes in the buffer

return self->inbuf_offset < self->inbuf_length;

}

STATIC void mp3file_update_inbuf_cb(void *self) {

mp3file_update_inbuf_always(self);

}

STATIC bool mp3file_update_inbuf_half(audiomp3_mp3file_obj_t *self) {

// If buffer is over half full, do nothing

if (self->inbuf_offset < self->inbuf_length / 2) {

return true;

}

return mp3file_update_inbuf_always(self);

}

#define READ_PTR(self) (self->inbuf + self->inbuf_offset)

#define BYTES_LEFT(self) (self->inbuf_length - self->inbuf_offset)

#define CONSUME(self, n) (self->inbuf_offset += n)

STATIC void mp3file_skip_id3v2(audiomp3_mp3file_obj_t *self) {

mp3file_update_inbuf_half(self);

if (BYTES_LEFT(self) < 10) {

return;

}

uint8_t *data = READ_PTR(self);

if (!(

data[0] == 'I' &&

data[1] == 'D' &&

data[2] == '3' &&

data[3] != 0xff &&

data[4] != 0xff &&

(data[5] & 0x1f) == 0 &&

(data[6] & 0x80) == 0 &&

(data[7] & 0x80) == 0 &&

(data[8] & 0x80) == 0 &&

(data[9] & 0x80) == 0)) {

return;

}

uint32_t size = (data[6] << 21) | (data[7] << 14) | (data[8] << 7) | (data[9]);

size += 10; // size excludes the "header" (but not the "extended header")

// First, deduct from size whatever is left in buffer

uint32_t to_consume = MIN(size, BYTES_LEFT(self));

CONSUME(self, to_consume);

size -= to_consume;

// Next, seek in the file after the header

f_lseek(&self->file->fp, f_tell(&self->file->fp) + size);

return;

}

STATIC bool mp3file_find_sync_word(audiomp3_mp3file_obj_t *self) {

do {

mp3file_update_inbuf_half(self);

int offset = MP3FindSyncWord(READ_PTR(self), BYTES_LEFT(self));

if (offset >= 0) {

CONSUME(self, offset);

mp3file_update_inbuf_half(self);

return true;

}

CONSUME(self, MAX(0, BYTES_LEFT(self) - 16));

} while (!self->eof);

return false;

}

STATIC bool mp3file_get_next_frame_info(audiomp3_mp3file_obj_t *self, MP3FrameInfo *fi) {

int err;

do {

err = MP3GetNextFrameInfo(self->decoder, fi, READ_PTR(self));

if (err == ERR_MP3_NONE) {

break;

}

CONSUME(self, 1);

mp3file_find_sync_word(self);

} while (!self->eof);

return err == ERR_MP3_NONE;

}

void common_hal_audiomp3_mp3file_construct(audiomp3_mp3file_obj_t *self,

pyb_file_obj_t *file,

uint8_t *buffer,

size_t buffer_size) {

// XXX Adafruit_MP3 uses a 2kB input buffer and two 4kB output buffers.

// for a whopping total of 10kB buffers (+mp3 decoder state and frame buffer)

// At 44kHz, that's 23ms of output audio data.

//

// We will choose a slightly different allocation strategy for the output:

// Make sure the buffers are sized exactly to match (a multiple of) the

// frame size; this is typically 2304 * 2 bytes, so a little bit bigger

// than the two 4kB output buffers, except that the alignment allows to

// never allocate that extra frame buffer.

self->inbuf_length = 2048;

self->inbuf_offset = self->inbuf_length;

self->inbuf = m_malloc(self->inbuf_length, false);

if (self->inbuf == NULL) {

common_hal_audiomp3_mp3file_deinit(self);

mp_raise_msg(&mp_type_MemoryError,

translate("Couldn't allocate input buffer"));

}

self->decoder = MP3InitDecoder();

if (self->decoder == NULL) {

common_hal_audiomp3_mp3file_deinit(self);

mp_raise_msg(&mp_type_MemoryError,

translate("Couldn't allocate decoder"));

}

if ((intptr_t)buffer & 1) {

buffer += 1;

buffer_size -= 1;

}

if (buffer_size >= 2 * MAX_BUFFER_LEN) {

self->buffers[0] = (int16_t *)(void *)buffer;

self->buffers[1] = (int16_t *)(void *)(buffer + MAX_BUFFER_LEN);

} else {

self->buffers[0] = m_malloc(MAX_BUFFER_LEN, false);

if (self->buffers[0] == NULL) {

common_hal_audiomp3_mp3file_deinit(self);

mp_raise_msg(&mp_type_MemoryError,

translate("Couldn't allocate first buffer"));

}

self->buffers[1] = m_malloc(MAX_BUFFER_LEN, false);

if (self->buffers[1] == NULL) {

common_hal_audiomp3_mp3file_deinit(self);

mp_raise_msg(&mp_type_MemoryError,

translate("Couldn't allocate second buffer"));

}

}

common_hal_audiomp3_mp3file_set_file(self, file);

}

void common_hal_audiomp3_mp3file_set_file(audiomp3_mp3file_obj_t *self, pyb_file_obj_t *file) {

background_callback_begin_critical_section();

self->file = file;

f_lseek(&self->file->fp, 0);

self->inbuf_offset = self->inbuf_length;

self->eof = 0;

self->other_channel = -1;

mp3file_update_inbuf_half(self);

mp3file_find_sync_word(self);

// It **SHOULD** not be necessary to do this; the buffer should be filled

// with fresh content before it is returned by get_buffer(). The fact that

// this is necessary to avoid a glitch at the start of playback of a second

// track using the same decoder object means there's still a bug in

// get_buffer() that I didn't understand.

memset(self->buffers[0], 0, MAX_BUFFER_LEN);

memset(self->buffers[1], 0, MAX_BUFFER_LEN);

MP3FrameInfo fi;

bool result = mp3file_get_next_frame_info(self, &fi);

background_callback_end_critical_section();

if (!result) {

mp_raise_msg(&mp_type_RuntimeError,

translate("Failed to parse MP3 file"));

}

self->sample_rate = fi.samprate;

self->channel_count = fi.nChans;

self->frame_buffer_size = fi.outputSamps * sizeof(int16_t);

self->len = 2 * self->frame_buffer_size;

}

void common_hal_audiomp3_mp3file_deinit(audiomp3_mp3file_obj_t *self) {

MP3FreeDecoder(self->decoder);

self->decoder = NULL;

self->inbuf = NULL;

self->buffers[0] = NULL;

self->buffers[1] = NULL;

self->file = NULL;

}

bool common_hal_audiomp3_mp3file_deinited(audiomp3_mp3file_obj_t *self) {

return self->buffers[0] == NULL;

}

uint32_t common_hal_audiomp3_mp3file_get_sample_rate(audiomp3_mp3file_obj_t *self) {

return self->sample_rate;

}

void common_hal_audiomp3_mp3file_set_sample_rate(audiomp3_mp3file_obj_t *self,

uint32_t sample_rate) {

self->sample_rate = sample_rate;

}

uint8_t common_hal_audiomp3_mp3file_get_bits_per_sample(audiomp3_mp3file_obj_t *self) {

return 16;

}

uint8_t common_hal_audiomp3_mp3file_get_channel_count(audiomp3_mp3file_obj_t *self) {

return self->channel_count;

}

bool audiomp3_mp3file_samples_signed(audiomp3_mp3file_obj_t *self) {

return true;

}

void audiomp3_mp3file_reset_buffer(audiomp3_mp3file_obj_t *self,

bool single_channel,

uint8_t channel) {

if (single_channel && channel == 1) {

return;

}

// We don't reset the buffer index in case we're looping and we have an odd number of buffer

// loads

background_callback_begin_critical_section();

f_lseek(&self->file->fp, 0);

self->inbuf_offset = self->inbuf_length;

self->eof = 0;

self->other_channel = -1;

mp3file_update_inbuf_half(self);

mp3file_skip_id3v2(self);

mp3file_find_sync_word(self);

background_callback_end_critical_section();

}

audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t *self,

uint32_t *buffer_length) {

if (!self->inbuf) {

return GET_BUFFER_ERROR;

}

if (!single_channel) {

channel = 0;

}

*buffer_length = self->frame_buffer_size;

if (channel == self->other_channel) {

*bufptr = (uint8_t *)(self->buffers[self->other_buffer_index] + channel);

self->other_channel = -1;

return GET_BUFFER_MORE_DATA;

}

self->buffer_index = !self->buffer_index;

self->other_channel = 1 - channel;

self->other_buffer_index = self->buffer_index;

int16_t *buffer = (int16_t *)(void *)self->buffers[self->buffer_index];

*bufptr = (uint8_t *)buffer;

mp3file_skip_id3v2(self);

if (!mp3file_find_sync_word(self)) {

return self->eof ? GET_BUFFER_DONE : GET_BUFFER_ERROR;

}

int bytes_left = BYTES_LEFT(self);

uint8_t *inbuf = READ_PTR(self);

int err = MP3Decode(self->decoder, &inbuf, &bytes_left, buffer, 0);

CONSUME(self, BYTES_LEFT(self) - bytes_left);

if (self->inbuf_offset >= 512) {

background_callback_add(

&self->inbuf_fill_cb,

mp3file_update_inbuf_cb,

self);

}

if (err) {

return GET_BUFFER_DONE;

}

return GET_BUFFER_MORE_DATA;

}

void audiomp3_mp3file_get_buffer_structure(audiomp3_mp3file_obj_t *self, bool single_channel,

bool *single_buffer, bool *samples_signed,

uint32_t *max_buffer_length, uint8_t *spacing) {

*single_buffer = false;

*samples_signed = true;

*max_buffer_length = self->frame_buffer_size;

if (single_channel) {

*spacing = self->channel_count;

} else {

*spacing = 1;

}

}

float common_hal_audiomp3_mp3file_get_rms_level(audiomp3_mp3file_obj_t *self) {

float sumsq = 0.f;

// Assumes no DC component to the audio. Is that a safe assumption?

int16_t *buffer = (int16_t *)(void *)self->buffers[self->buffer_index];

for (size_t i = 0; i < self->frame_buffer_size / sizeof(int16_t); i++) {

sumsq += (float)buffer[i] * buffer[i];

}

return sqrtf(sumsq) / (self->frame_buffer_size / sizeof(int16_t));

}