circuitpython/shared-module/audioio/WaveFile.c
Scott Shawcroft 76008ce304
Introduce audioio.Mixer which can mix multiple audio samples
to produce a single sample.

Only works with 16 bit samples on the M4.

Fixes #987
2018-10-05 15:12:23 -07:00

261 lines
9.1 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
*
* 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/audioio/WaveFile.h"
#include <stdint.h>
#include <string.h>
#include "py/mperrno.h"
#include "py/runtime.h"
#include "shared-module/audioio/WaveFile.h"
#include "supervisor/shared/translate.h"
struct wave_format_chunk {
uint16_t audio_format;
uint16_t num_channels;
uint32_t sample_rate;
uint32_t byte_rate;
uint16_t block_align;
uint16_t bits_per_sample;
uint16_t extra_params; // Assumed to be zero below.
};
void common_hal_audioio_wavefile_construct(audioio_wavefile_obj_t* self,
pyb_file_obj_t* file) {
// Load the wave
self->file = file;
uint8_t chunk_header[16];
f_rewind(&self->file->fp);
UINT bytes_read;
if (f_read(&self->file->fp, chunk_header, 16, &bytes_read) != FR_OK) {
mp_raise_OSError(MP_EIO);
}
if (bytes_read != 16 ||
memcmp(chunk_header, "RIFF", 4) != 0 ||
memcmp(chunk_header + 8, "WAVEfmt ", 8) != 0) {
mp_raise_ValueError(translate("Invalid wave file"));
}
uint32_t format_size;
if (f_read(&self->file->fp, &format_size, 4, &bytes_read) != FR_OK) {
mp_raise_OSError(MP_EIO);
}
if (bytes_read != 4 ||
format_size > sizeof(struct wave_format_chunk)) {
mp_raise_ValueError(translate("Invalid format chunk size"));
}
struct wave_format_chunk format;
if (f_read(&self->file->fp, &format, format_size, &bytes_read) != FR_OK) {
mp_raise_OSError(MP_EIO);
}
if (bytes_read != format_size) {
}
if (format.audio_format != 1 ||
format.num_channels > 2 ||
format.bits_per_sample > 16 ||
(format_size == 18 &&
format.extra_params != 0)) {
mp_raise_ValueError(translate("Unsupported format"));
}
// Get the sample_rate
self->sample_rate = format.sample_rate;
self->len = 256;
self->channel_count = format.num_channels;
self->bits_per_sample = format.bits_per_sample;
// TODO(tannewt): Skip any extra chunks that occur before the data section.
uint8_t data_tag[4];
if (f_read(&self->file->fp, &data_tag, 4, &bytes_read) != FR_OK) {
mp_raise_OSError(MP_EIO);
}
if (bytes_read != 4 ||
memcmp((uint8_t *) data_tag, "data", 4) != 0) {
mp_raise_ValueError(translate("Data chunk must follow fmt chunk"));
}
uint32_t data_length;
if (f_read(&self->file->fp, &data_length, 4, &bytes_read) != FR_OK) {
mp_raise_OSError(MP_EIO);
}
if (bytes_read != 4) {
mp_raise_ValueError(translate("Invalid file"));
}
self->file_length = data_length;
self->data_start = self->file->fp.fptr;
// Try to allocate two buffers, one will be loaded from file and the other
// DMAed to DAC.
self->buffer = m_malloc(self->len, false);
if (self->buffer == NULL) {
common_hal_audioio_wavefile_deinit(self);
mp_raise_msg(&mp_type_MemoryError, translate("Couldn't allocate first buffer"));
}
self->second_buffer = m_malloc(self->len, false);
if (self->second_buffer == NULL) {
common_hal_audioio_wavefile_deinit(self);
mp_raise_msg(&mp_type_MemoryError, translate("Couldn't allocate second buffer"));
}
}
void common_hal_audioio_wavefile_deinit(audioio_wavefile_obj_t* self) {
self->buffer = NULL;
}
bool common_hal_audioio_wavefile_deinited(audioio_wavefile_obj_t* self) {
return self->buffer == NULL;
}
uint32_t common_hal_audioio_wavefile_get_sample_rate(audioio_wavefile_obj_t* self) {
return self->sample_rate;
}
void common_hal_audioio_wavefile_set_sample_rate(audioio_wavefile_obj_t* self,
uint32_t sample_rate) {
self->sample_rate = sample_rate;
}
bool audioio_wavefile_samples_signed(audioio_wavefile_obj_t* self) {
return self->bits_per_sample > 8;
}
uint32_t audioio_wavefile_max_buffer_length(audioio_wavefile_obj_t* self) {
return 512;
}
void audioio_wavefile_reset_buffer(audioio_wavefile_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
self->bytes_remaining = self->file_length;
f_lseek(&self->file->fp, self->data_start);
self->read_count = 0;
self->left_read_count = 0;
self->right_read_count = 0;
}
audioio_get_buffer_result_t audioio_wavefile_get_buffer(audioio_wavefile_obj_t* self,
bool single_channel,
uint8_t channel,
uint8_t** buffer,
uint32_t* buffer_length) {
if (!single_channel) {
channel = 0;
}
uint32_t channel_read_count = self->left_read_count;
if (channel == 1) {
channel_read_count = self->right_read_count;
}
bool need_more_data = self->read_count == channel_read_count;
if (self->bytes_remaining == 0 && need_more_data) {
*buffer = NULL;
*buffer_length = 0;
return GET_BUFFER_DONE;
}
if (need_more_data) {
uint16_t num_bytes_to_load = self->len;
if (num_bytes_to_load > self->bytes_remaining) {
num_bytes_to_load = self->bytes_remaining;
}
UINT length_read;
if (self->buffer_index % 2 == 1) {
*buffer = self->second_buffer;
} else {
*buffer = self->buffer;
}
if (f_read(&self->file->fp, *buffer, num_bytes_to_load, &length_read) != FR_OK) {
return GET_BUFFER_ERROR;
}
self->bytes_remaining -= length_read;
// Pad the last buffer to word align it.
if (self->bytes_remaining == 0 && length_read % sizeof(uint32_t) != 0) {
uint32_t pad = length_read % sizeof(uint32_t);
length_read += pad;
if (self->bits_per_sample == 8) {
for (uint32_t i = 0; i < pad; i++) {
((uint8_t*) (*buffer))[length_read / sizeof(uint8_t) - i - 1] = 0x80;
}
} else if (self->bits_per_sample == 16) {
// We know the buffer is aligned because we allocated it onto the heap ourselves.
#pragma GCC diagnostic push
#pragma GCC diagnostic ignored "-Wcast-align"
((int16_t*) (*buffer))[length_read / sizeof(int16_t) - 1] = 0;
#pragma GCC diagnostic pop
}
}
*buffer_length = length_read;
if (self->buffer_index % 2 == 1) {
self->second_buffer_length = length_read;
} else {
self->buffer_length = length_read;
}
self->buffer_index += 1;
self->read_count += 1;
}
uint32_t buffers_back = self->read_count - 1 - channel_read_count;
if ((self->buffer_index - buffers_back) % 2 == 0) {
*buffer = self->second_buffer;
*buffer_length = self->second_buffer_length;
} else {
*buffer = self->buffer;
*buffer_length = self->buffer_length;
}
if (channel == 0) {
self->left_read_count += 1;
} else if (channel == 1) {
self->right_read_count += 1;
*buffer = *buffer + self->bits_per_sample / 8;
}
return self->bytes_remaining == 0 ? GET_BUFFER_DONE : GET_BUFFER_MORE_DATA;
}
void audioio_wavefile_get_buffer_structure(audioio_wavefile_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 = self->bits_per_sample > 8;
*max_buffer_length = 512;
if (single_channel) {
*spacing = self->channel_count;
} else {
*spacing = 1;
}
}