circuitpython/shared-module/audiomp3/MP3Decoder.c

353 lines
12 KiB
C

/*
* This file is part of the Micro Python project, http://micropython.org/
*
* The MIT License (MIT)
*
* Copyright (c) 2018 Scott Shawcroft for Adafruit Industries
* Copyright (c) 2019 Jeff Epler for Adafruit Industries
*
* Permission is hereby granted, free of charge, to any person obtaining a copy
* of this software and associated documentation files (the "Software"), to deal
* in the Software without restriction, including without limitation the rights
* to use, copy, modify, merge, publish, distribute, sublicense, and/or sell
* copies of the Software, and to permit persons to whom the Software is
* furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
* OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN
* THE SOFTWARE.
*/
#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 "lib/mp3/src/mp3common.h"
#define MAX_BUFFER_LEN (MAX_NSAMP * MAX_NGRAN * MAX_NCHAN * sizeof(int16_t))
/** Fill the input buffer if it is less than half full.
*
* Returns true if the input buffer contains any useful data,
* false otherwise. (The input buffer will be padded to the end with
* 0 bytes, which do not interfere with MP3 decoding)
*
* Raises OSError if f_read fails.
*
* Sets self->eof if any read of the file returns 0 bytes
*/
STATIC bool mp3file_update_inbuf(audiomp3_mp3file_obj_t* self) {
// If buffer is over half full, do nothing
if (self->inbuf_offset < self->inbuf_length/2) return true;
// 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;
}
#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)
// http://id3.org/d3v2.3.0
// http://id3.org/id3v2.3.0
STATIC void mp3file_skip_id3v2(audiomp3_mp3file_obj_t* self) {
mp3file_update_inbuf(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;
}
/* If a sync word can be found, advance to it and return true. Otherwise,
* return false.
*/
STATIC bool mp3file_find_sync_word(audiomp3_mp3file_obj_t* self) {
do {
mp3file_update_inbuf(self);
int offset = MP3FindSyncWord(READ_PTR(self), BYTES_LEFT(self));
if (offset >= 0) {
CONSUME(self, offset);
mp3file_update_inbuf(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) {
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(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;
if(!mp3file_get_next_frame_info(self, &fi)) {
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
f_lseek(&self->file->fp, 0);
self->inbuf_offset = self->inbuf_length;
self->eof = 0;
self->other_channel = -1;
mp3file_update_inbuf(self);
mp3file_skip_id3v2(self);
mp3file_find_sync_word(self);
}
audioio_get_buffer_result_t audiomp3_mp3file_get_buffer(audiomp3_mp3file_obj_t* self,
bool single_channel,
uint8_t channel,
uint8_t** bufptr,
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 (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));
}