circuitpython/shared-module/audiocore/WaveFile.c
Radomir Dopieralski 9907e3fa28 Allow to specify pre-allocated buffer in audicore.WaveFile
It lets us re-use the same buffer for playing multiple files.
This also allows us to control the size of the buffer. Half of the
buffer will be used for the fist, and half for the second internal
buffer.
2019-08-03 14:15:19 +02:00

280 lines
9.7 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/audiocore/WaveFile.h"
#include <stdint.h>
#include <string.h>
#include "py/mperrno.h"
#include "py/runtime.h"
#include "shared-module/audiocore/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,
uint8_t *buffer,
size_t buffer_size) {
// 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->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.
if (buffer_size) {
self->len = buffer_size / 2;
self->buffer = buffer;
self->second_buffer = buffer + self->len;
} else {
self->len = 256;
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;
self->second_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;
}
uint8_t common_hal_audioio_wavefile_get_bits_per_sample(audioio_wavefile_obj_t* self) {
return self->bits_per_sample;
}
uint8_t common_hal_audioio_wavefile_get_channel_count(audioio_wavefile_obj_t* self) {
return self->channel_count;
}
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;
}
}